Try this Quiz and Test Your VoIP and SIP Knowledge

Someone sent us a link to this VoIP Fundamentals Quiz on SearchUnifiedCommunications by TechTarget, because, you know, we’re totally into VoIP.  We weren’t surprised that a publication created a quiz about VoIP and SIP, but we were surprised by how technical and tricky it is.  Give it a whirl, then come back for our take on each of the answers.

Try the Quiz at SearchUnifiedCommunications

How’d you do?  We agree that their answers are accurate, but we’ve got a bit more detail and some SIP specific thoughts to add to the questions.  (Spoiler alert: This is commentary, not a repeat of the quiz, so if you didn’t take the quiz yet, don’t get upset if we take all the fun out of it.)

Question 1

The first question asked you to identify one non-VoIP protocol among 3 VoIP protocols.  The target answer is SS7, which is indeed a non-IP protocol used by the PSTN. The other choices were SIP, H.225 and H.225 RAS.  Of course we know that SIP is a VoIP protocol (read here about how all SIP is VoIP, but not all VoIP is SIP), as are H.225 and H.225 RAS.  If you know that RAS stands for Registration, Admission and Status, you probably got this one right.

Question 2

This question is about the factors that influence voice quality in VoIP calls.  Choices included bandwidth, latency, codec support, the type of echo cancellation in use and weather conditions.  And as SearchUnifiedCommunications rightly concludes, it’s an all of the above situation.  Another factor that they don’t mention is the tier of the upstream network.  In order to insure high quality for our clients, for example, we use only Tier-1 upstream carriers.

Question 3

The subject this time are the symptoms of VoIP issues.  Once again, it’s an all of the above. The potential problem include, voice working for only one party, unexpected sounds, irregular drop-out intervals and echo.  Echo is the worst.  In order to make sure that the VoIP or SIP solution you select doesn’t drive you nuts with any of these problems, we recommend offering a SIP provider that offers a free trial period.

Question 4

This time the attention is on the bandwidth required for VoIP.  Although the number they selected, 90 Kbps is a good standard round number, there is some variation, depending on the codec used.  Our service uses the G.722 codec that requires about 85kbps.  We also allow, but in most cases do not recommend the G.729 compressed codec that only requires 35kbps.  The compression, however, does impact quality so the bandwidth trade off needs to be worth it.

Question 5

This one asks about the mean opinion score (MOS) used to grade perceived call quality for VoIP. We won’t repeat all the possible incorrect choices, but the MOS score is an average value for call quality based on a five-point scale of values 1 through 5, where 1 indicates communication is impossible and 5 is like face-to-face interaction.  The MOS scoring method has been in use for decades.  As the inclusion of the word “opinion” implies, it is a subjective measurement and varying test conditions, test subjects and other uncontrollable variables make it an interesting, but not necessarily conclusive, assessment of audio quality.  We prefer the try it for yourself method.

Question 6

Question 6 points out that MOS does not measure latency directly, although the symptoms of latency, if present, will almost certainly result in a lower MOS score.  Unlike “audio quality,” which can only be measured by human perception, data latency can be measured.  We believe that a quality SIP experience requires latency of less than 150 milliseconds.

Question 7

Speaking of latency, question 7 tests whether you know that echo is a symptom of connection latency.  We don’t have much to add here except that latency is bad news for IP calls.

Question 8

We’ll forgive you if you missed question 8.  It asks, “Which of the following switch or router capabilities can help improve VoIP call quality on a LAN?” The options are:

• Use MPLS to speed general traffic handling
• Enforce Quality of Service (QoS) policies that give preferential treatment to VoIP packets
• Use Differentiated Services (DiffServ) to give preferential treatment to VoIP packets
• Use pre-emptive packet drop techniques to limit packet delays for VoIP traffic

The answer they were looking for was, “Enforce Quality of Service,” and we agree, that’s exactly what you should do, but the others are technically possible, just not very common or necessarily effective against VoIP problems.  So, if you missed this one, give yourself a mulligan.

Question 9

Question 9 made us realize that we don’t talk very much about IP phones in this blog. We’ll devote some space to that soon. But SearchUnifiedCommunications makes a good point with this question that support for unified communications like instant messaging, video, and visual voicemail are important features of modern IP enabled handsets.

Question 10

The final question is about the digital signal processor (DSP) which is a core component of VoIP that handles compression, conversion, quantization and encoding.

The quiz credits telecommunications expert Ed Tittle.  Ed, our hats are off to you. If you missed every question on the quiz, don’t panic. You don’t need to know the answer to any of them to effectively use and deploy a SIP trunk. Our experts eat and breath this stuff just so you don’t have to. But, if you got them all right kudos! Drop us a line and we’ll send you instructions for the secret handshake.