Archives for SIP

Choosing a Proper “Dialer Friendly” SIP Provider

When people that are new to setting up a outbound call center choose a SIP provider for a Vicidial (or similar) based call center, they just choose a SIP provider based purely on the lowest cost per outbound minute.  But there are MANY more factors to consider.  With many of our customers, we have consulted with them after they have signed a multi-year contract only to find that the provider they have selected won’t fit their needs.

In the following article, we will outline a few of the factors you need to consider when choosing a SIP provider for dialtone for your outbound call center.

True Cost Per Minute

I realize that in the first paragraph it may seem like I said this wasn’t the prime factor, but in fact it IS an important factor.  HOWEVER, there’s more to it than just raw cost/minute.  An average cost for an outbound call in the contiguous United States (i.e. not Alaska and Hawaii) is $0.015/minute.  If you are calculating all your outbound dialing costs on that figure, it would be simple, however you can get better rates for dialing with what is known as “rate deck” plans.  That is to say, calling a certain state might cost you $0.008/minute, but calling another state would cost you $0.017/minute.  Depending on your target regions – you might be able to save money on your outbound costs by doing a little research.

Additionally, the billing method is important.  Some carriers will bill you for the first 30 seconds of a call, then bill you in increments after those first 30 seconds.  Others will bill you purely in 6 second increments.  If your dialing patterns give you a lot of short calls interspersed with a few longer calls (as most dialing traffic typically is), this is most likely the best option.

Short Call Duration Penalty

One little billing fee most customers are not aware of that exists with many SIP carriers, is what is known as a “Short Call Duration Penalty”.  Dialer SIP traffic is unusual.  Phone companies don’t make a lot of money from it because the bulk of the calls don’t connect and thus can’t be billed directly per minute (unless its written that way in the contract).  For that reason, many non-dialer-friendly carriers impost a the short call duration penalty when the percentage of calls that don’t connect versus those that do goes over a certain threshold.  This can add up quickly for larger call centers.

Calls Per Second Capacity

Here’s an area that people who don’t do their research can get stuck in very quickly.  An autodialer is designed to send out lots of calls at the same time, and with some dialers, there can be hundreds of calls going on at once, with more being added each time a call gets disconnected.  So the maximum calls-per-second, or CPS, rate is important for how quickly a dialer can recover from dropped or disconnected calls thus keeping agents on the phone and a call center productive.

Maximum CPS starts around 1-2 CPS and goes up from there.  In call centers with simultaneous calls in the hundreds, this number should be greater than 10 CPS.  This is where the smaller carriers fall short.  Most smaller SIP carriers can only handle a MAXIMUM of 5-10 CPS (although of course this varies from carrier to carrier).

Call Setup Time

One of the most crucial metrics for a outbound dialing is the time it takes from when the dialer initiates the call to when the call starts ringing on the other end.  This may not seem important, but it affects the outbound call center metric known as “drop rate”.  If the call takes too long to set up, the dialer may treat it as a failed call, and mark it as “dropped” and move on to the next call.  The problem is that this also affects a parameter known as “contact rate” that affects how fast the dialer has to run in order to keep agents on the phone.  What does all that mean?  Essentially, if the SIP carrier you choose has too high of a call setup time, it will take longer to make sales or cost more to make sales because you have to make more calls than with a carrier that has a lower call setup time.

Often, this occurs with carriers who claim they can give you the “best possible rate” and come back with an outrageously cheap price (I’ve seen quotes of $0.004/minute – that’s less than a half a penny per minute).  These carriers can do some funky stuff with the traffic (like send it offshore, and route it back through different carriers, or use funky routing, etc) to get these rates, but consequently they will have much higher call setup times.

The best way to get the best call setup time, is to go with a recommendation from a currently operating call center.  Or contact us so we can give you the benefit of our expertise with various SIP carriers friendly to dialer traffic and help you select one.

Inbound Number Availability

Surprising as it may sound, there are in fact a few “pro-dialer” carriers that meet the above criteria swimmingly, but do not have the ability to provide numbers that can be called back and routed back into your call center.  This is a major problem because the large majority of outbound calls go unanswered and if a valid number is on the caller ID, the person who was called could call back in and a possible sale could be made.

Focusing on the quality and cost of the outbound calls is important, it is just as important that the people you call can call you back, and that your SIP provider will facilitate that.

Number of Supporting “Backbone” Carriers

Let’s face it, there are literally hundreds, if not thousands, of SIP providers available today.  Each are competing for a share of the dialer market.  Some smaller carriers though are “mom-and-pop” operations.  While there is nothing inherently wrong with that, few have the capacity to hand the traffic that dialers generate.  Most often a small SIP provider will have a single connection to a major carrier.  The upside is that they can usually get pretty good rates to pass on, the downside is that they can only handle so much traffic.  Plus, if that one connection to their provider goes down, that’s it.  All their clients loose their ability to make calls.

Single Carrier SIP Provider

SIP Provider With a Single Carrier

Some of the better carriers have multiple connections to other providers to avoid this problem.  The problem is that many of the carriers they connect to, aren’t what are known as “backbone” carriers (i.e. the “500 pound gorillas of telecom – Level 3, Verizon, AT&T), but instead are just other more regional carriers.  They can handle more volume, and are more redundant, but they do have a higher cost factor, and they typically have a much higher call setup time than a smaller carrier.

The best carriers have multiple, direct connections to the backbone carriers (as shown in the image below).  They can maintain a high call volume, keep rates relatively low (although admittedly will be higher than the “mom-and-pop” operations), have extremely low call setup times, and have multiple routes for redundancy.

multi homed SIP provider

Multi-Homed SIP Provider

Local Caller ID Presence Numbers

These days, when sales calls are made, the person on the ringing end of the phone can usually see who is calling by use of caller ID.  But, as most of us who have received telemarketing calls know, if its a toll free number (800, 844, 888, etc), generally speaking it is a telemarketer or someone you don’t want to talk to.  But sales call centers need to reach people to make sales.  So what they often do is buy “local” DID numbers that point back to their call center from anywhere in the country.  Then, the call center will setup their dialer so that when they call a number, the dialer will send out a caller ID that is “local” to the caller.  For example, if the customer’s number is 909-111-1111, when the dialer dials that number, it will recognize the 909 area code, look up in its database of DID’s, and send out the caller ID of a phone number that points back to the dialer (say 909-222-2222 for example).  The caller is more likely to pick up, or if they miss the call, call them back. Using this tactic greatly increases contact rates.

NOTE: Vicidial is one dialer that has this capability

The issue is that most carriers require that you purchase INDIVIDUAL phone numbers in EACH area code.  With over 290 area codes in the US, that’s a lot of phone numbers to buy one at a time. Not only that, but people eventually will recognize these local numbers as telemarketing numbers and start blocking them, so you would have to buy new numbers every few months.  So while it has benefits, if not done properly this approach can be costly.

A good dialer SIP provider will have a “local presence” DID package (name of the package varies by carrier) that will provide local phone numbers in each area code.  Some providers even include FREE minutes with each number.  Additionally, some providers will allow you to “refresh” this package a few times a year to get new numbers to keep them working.

Summary

When choosing a SIP provider for your dialer call center, remember that cost is only one factor.  Multiple factors contribute to the proper interoperability of a SIP provider and a call centers and all these factors should be considered when making your choice.

For more information on SIP providers, Vicidial, call center design and implementation or help choosing a SIP provider, please contact us at info@verterion.com

Asterisk Call Quality Troubleshooting

Excellent call quality with any phone system is crucial.  While businesses these days do run on email and web, the telephone is STILL the lifeline of the business. Asterisk as a call processing engine for a PBX is well designed and very efficient, so it has many features that can be used to mitigate the most common problems.  In this article we’ll briefly discuss the most common call quality issues that occur with IP PBX’s and how Asterisk-based systems can deal with him.

NOTE: These tips primarily deal with Internet-based SIP dial tone, except where noted.

Jitter

This is arguably the most common issue with any IP-based PBX.  It especially occurs at some point on systems with Internet-based SIP trunks for their dial tone.  Jitter sounds like one of two things: Either “interruptions” in the audio, or “underwater sounding voices”.  The issue is caused by inconsistencies in the delay between data packets between two endpoints.  Most people, incorrectly, assume that they don’t have enough Internet bandwidth.  This is generally not the source of the problem (unless the Internet is painfully underpowered for the amount of calls going through at the same time – i.e. dialer traffic).  It has more to do with the QUALITY of the Internet than the QUANTITY of bandwidth.  You can even have this problem on a LAN, where there is NO Internet connection and plenty of available bandwidth, but the LAN is congested or of poor performance.

With a poor quality Internet connection where there are inconsistent delays between packets, there isn’t a lot you can do with the provider.  Once it leaves your network for the Internet, its out of your control.  HOWEVER, Asterisk (and other IP PBX’s) have a function called a Jitter Buffer. A Jitter Buffer takes a portion of the audio and “buffers” it (stores it in memory briefly) before sending or while receiving.  This has the benefit of removing or eliminating the jitter.

Occasionally a possible cause of jitter is processor overload on the firewall or switches.  You can use your firewall or switch management tools to see if they are overloaded, then either remove some of the load, or upgrade them and see if the jitter is reduced and call quality improves.

One Way Audio

One way audio is a particular problem  that is usually seen when a PBX is first installed.   What it sounds like should be pretty obvious, based on the name.  Only one party will hear the audio portion of the call.  The call will ring, but when the destination party picks up, the receiver hears the call, but can’t talk back to the originator.

This problem is due to a misconfiguration of the Asterisk PBX, the firewall between it and the provider, or both.  When a firewall sits between an Asterisk PBX and an Internet SIP provider, all packets go through Network Address Translation (NAT).  In other words, the private IP address in the network packet is replaced with the public IP address of the firewall when the packet travels out, and the reverse happens on the return.

The issue specific to SIP is that a SIP packet is unique in that it has TWO places for the IP address, and a firewall typically isn’t smart enough to look for the second IP address (located in the SIP header of the packet).  So when a SIP provider tries to send its traffic back to the firewall that sent it – it is going to try to send it to the IP address listed in the SIP header, which is a private LAN address can’t be routed over the Internet.

Some firewalls have a function called an Application Layer Gateway (ALG) – and it goes by different names by different manufacturers (SIP ALG, SIP Fixup, SIP-NAT, and so on), but they all basically do the same thing – look for SIP packets and get involved to do the SECOND IP address NATing.

Unfortunately, most firewalls do this incorrectly with Asterisk-based systems.  Asterisk has a function built into its SIP configuration for automatically taking care of this issue.  The function uses 3 parameters in it’s sip.conf configuration file (the location and use of which depends on which version and implementation of Asterisk you are using).  These parameters are:

  • NAT=yes
  • localnet=<local network definition>
  • externip=<external NAT’ed IP address of the PBX>

When these three parameters are configured correctly, the PBX automatically takes care of the NAT substitution for packets destined for Internet hosts.

NOTE: Because SIP ALG’s generally aren’t smart enough or often don’t work properly with Asterisk – we highly recommend turning them OFF and properly configuring Asterisk as described

Echo

Troubleshooting echo issues isn’t typically an issue with IP PBX systems using SIP providers, it is more commonly seen with systems that use more traditional connections to the telephone company.  Connections such as PRI, or more commonly, analog lines.

Echo, as you may guess, is simply a repeat of some portion of the audio back to either speaker.  There are two typical causes.  The first is easiest to solve.  Often, someone will be calling on a speaker phone with the volume turned up significantly and a portion of the audio is “feeding back” into the microphone.  The person on the other end of the call will hear a portion of the audio delayed by a few milliseconds.  The solution for this issue is straightforward, simply turn down the speaker volume and see if the quality improves (or pickup the handset and see if the problem goes away completely).

The more complicated cause is what is known as a “hot” volume on a trunk.  That means that the incoming voltage is too high for the interface card to handle and it must be adjusted down to compensate.  If the incoming voltage is too high, the caller from the outside will receive a portion of the signal echoing back.  If the outgoing voltage is too high, the caller from the inside will receive a portion of the signal echoing back.  If both are too high, both parties may receive echos.

Asterisk has the ability to adjust “gain” settings for its interface cards that can reduce or eliminate this echo if this is truly the source.  The parameters are “txgain” and “rxgain” located in the DAHDI or ZAPTEL configuration settings (depending on the technology used).

Some interface cards also have built in hardware to automatically eliminate echo on problematic lines.  If you are using interface cards without automatic echo cancelling, consider replacing them with cards with automatic echo cancelling cards.  Be warned, however, they are more expensive than their non-echo cancelling counterparts.

It is important to note that the amount of echo is also dependent on the volume of the person speaking.  If the echo problem is borderline, and the person speaking starts talking louder, then the echo may begin at that point because it has passed the threshold.

Summary

This article should give you a few items to look at when troubleshooting call quality issues on your Asterisk-based system (as well as any IP-PBX for that matter). For assistance with eliminating any of your call quality issues, feel free to contact us or email us at info@verterion.com

WP2Social Auto Publish Powered By : XYZScripts.com