Packet Voice Quality Problems and Answers

Almost all packet-based voice quality issues are attributable to some type of degradation on the packet network that the voice traffic's RTP stream traverses. Voice traffic brings to light network problems that might otherwise go unnoticed when just carrying normal data traffic. This is because in voice compressor-decompressors (CODECs) packet loss and variable delay in the IP telephony network must be minimized. Let's explore some common network issues that result in poor voice quality and what you can do about it:

Packet Drops

Packet-based telephony demands that speech packets find their destination within a predictable amount of time. There is very little tolerance for them to be dropped somewhere along the way from the source to the destination. In a properly designed network with Quality of Service (QoS) provisioning in place, packet loss should be nearly zero. All voice CODECs can tolerate degrees of packet loss without adversely affecting voice quality. Upon detecting a missing packet, the CODEC decoder on the receiving device makes a best guess as to what the waveform during the missing period of time should have been. Most CODECs can tolerate up to five percent random packet loss without noticeable voice quality degradation. This assumes that the five percent of packets being lost are not being lost at the same time, but rather are randomly dropped in groups of one or two packets. However, losing multiple simultaneous packets, even as a low percentage of total packets, can cause noticeable voice quality problems.

Note: You should design your network for zero packet loss for packets that are tagged as voice packets. A converged voice/data network should be engineered to ensure that only a specific number of calls are allowed over a limited-bandwidth link. You should guarantee the bandwidth for those calls by giving priority treatment to voice traffic over all other traffic.

There are various tools and procedures that you can use to determine whether you are experiencing packet loss in your network and where in the network the packets are getting dropped.

1. Examine IP phone statistics:

• If you are troubleshooting at the phone experiencing the problem, access the phone statistics by pressing the help (i or ?) button on the IP phone twice in quick succession during an active call.
• If you are working with a remote user, open a web browser on your computer and enter the IP address of the user's phone. During an active call, choose the Streaming Statistics > Stream 1 options from the display.

2. Examine the counters RxDisc and RxLost shown on the IP phone (or Rcvr Lost Packets if you are viewing the statistics remotely using a web browser).

• RxLost measures the number of packets that were never received because they were dropped in the network somewhere. By detecting a missing RTP sequence number, the IP phone can determine that a packet has been lost.
• RxDisc corresponds to packets that were received but were discarded because they could not be used at the time they arrived. RxDisc can come from an out-of-order packet or a packet that arrived too late. 

3. If either of these two counters increments, you should investigate to learn why packets are being lost or discarded.

Regardless of how low your packet loss is, if it is not zero, you should investigate the root cause. It just might be a sign of a bigger problem that will get worse with higher call volume. Always remember that packet loss can be occurring at any layer of the OSI-like transmission model, so be sure to check for all layers possible in each hop. For example, if there is a Frame Relay connection over a T1 between two sites, you should:

• Make certain that there are no errors at the physical layer on the T1.
• Determine if you are exceeding your committed information rate (CIR) on the Frame Relay connection.
• Verify that you are not dropping the packets at the IP layer because you are exceeding your buffer sizes.
• Check that you have your QoS improperly configured.
• Ensure that your service provider not only guarantees packet delivery but also guarantees a low-jitter link. Some service providers may tell you that they do not provide a CIR but guarantee that they will not drop any packets. In a voice environment, delay is as important as packet loss. Many service providers' switches can buffer a large amount of data, thereby causing a large amount of jitter.

Another common cause of drops in an Ethernet environment is a duplex mismatch, when one side of a connection is set to full duplex and the other side is set to half duplex. To determine if this is the case, perform the following steps:

1. Check all the switch ports through which a given call must travel and ensure that there are no alignment or frame check sequence (FCS) errors. Poor cabling or connectors can also contribute to such errors; however, duplex mismatches are a far more common cause of this kind of problem.
2. Examine each link between the two endpoints that are experiencing packet loss and verify that the speed and duplex settings match on either side.

Although duplex mismatches are responsible for a large number of packet loss problems, there are many other opportunities for packet loss in other places in the network as well. When voice traffic must traverse a WAN, there are several places to look. First, check each interface between the two endpoints, and look for packet loss. If you are seeing dropped packets on any interface, there is a good chance that you are oversubscribing the link. This could also be indicative of some other traffic that you are not expecting on your network. The best solution in this case is to take a trace to examine which traffic is congesting the link.

Hosted/Web-based analysis tools are invaluable in troubleshooting voice quality problems. With these tools, you can examine each packet in an RTP stream to see if packets are really being lost and where in the network they are being lost. With such a tool, perform the following steps:

1. Start at the endpoint that is experiencing the poor-quality audio where you suspect packet loss.
2. Take a trace of a poor-quality call and filter it so that it shows you only packets from the far end to the endpoint that is hearing the problem. The packets should be equally spaced, and the sequence numbers should be consecutive with no gaps.
3. If you are seeing all the packets in the trace, continue taking traces after each hop until you get a trace where packets are missing.
4. When you have isolated the point in the network where the packet loss is occurring, look for any counters on that device that might indicate where the packets are being lost.

Queuing Problems

Queuing delay can be a significant contributor to variable delay (jitter). When you have too much jitter end-to-end, you encounter voice quality problems. A voice sample that is delayed over the size of the receiving device's jitter buffer is no better than a packet that is dropped in the network because the delay still causes a noticeable break in the audio stream. In fact, high jitter is actually worse than a small amount of packet loss because most CODECs can compensate for small amounts of packet loss. The only way to compensate for high jitter is to make the jitter buffer larger, but as the jitter buffer gets larger, the voice stream is delayed longer in the jitter buffer. If the jitter buffer gets large enough such that the end-to-end delay is more than 200ms, the two parties on the call feel like the conversation is not interactive and start talking over each other.

Remember that every network device between the two endpoints involved in a call (switches, routers, firewalls, and so on) is a potential source of queuing or buffering delays. The ideal way to troubleshoot a problem in which the symptoms point to delayed or jittered packets is to use a hosted diagnostic tool at each network hop to see where the delay or jitter is being introduced.


Also remember that once you've made changes to your network elements and call paths that it is always appropriate to perform regression testing across your network with some type of automated voice quality test package. 

MOS [Mean Opinion Score] for High Definition or Wideband Telephony

Mean Opinion Score [MOS] is a scale from 1 to 5 indicating speech quality - 1 is bad and 5 is excellent. MOS test sessions comprise 15 to 25 people listening to speech files of good quality and of poor quality with impairments and scoring them subjectively. This subjective test process is specified in ITU-T P.800. In over 16 years where these tests have taken place, no statistically significant number of participants ever scored any speech recording as being excellent or 5.0. The highest score typically obtained in any test was 4.5.

High Definition or Wideband Telephony speech uses the new POLQA speech quality metric for objective protection of MOS. The old PESQ algorithm has been used for narrowband telephony since it was approved in 2000. It is desirable to use the same scale so that laboratories can compare new results for wideband telephony with their old PESQ database. However, the question of human expectation comes into play because all these objective measurements performed by computers must correlate or predict subjective experience. If you watch a video on your smart phone, you might consider the picture quality as being good. Your expectations are put in the context of the small screen and the convenience of the video being played on a handheld smartphone. If you would give you the same video on your brand-new expensive high-definition 1080P TV, you would be very disappointed even if the pixel resolution had been scaled to the 62 inches screen size. Your expectation of quality is tempered to the format in which you are viewing it.

Similarly with speech and audio. If you were to participate in a MOS test and invited into a studio where there were high fidelity speakers, orchestral classical music playing and told and asked to rate the quality of the High Definition speech you are about to hear, your expectations would be set high and you'd be more critical. You would score the audio lower than if you had been asked to rate the speech quality of your most recent cellular phone call.



POLQA offers two scales, the narrowband scale and the super wideband scale. Super wideband telephony reaches 14 kHz analog audio frequency. The narrowband focus scale maps directly onto the old desk scale and exploits the higher scores not given by test participants in narrowband tests.



• NB: Maximum MOS value 4.25

• WB: Maximum MOS value 4.5

• SWB: Maximum MOS value 4.75



So a score of 4.5, on the narrowband POLQA scale is experimentally the best value you will ever obtain with wideband telephony equipment. You could conceivably measure a MOS value of 4.75 if you were measuring super wideband equipment.


In future years, the industry will migrate exclusively to using the super wideband POLQA scale as soon as users' expectations always expect high-definition or hi-fi quality to the communications audio.

The picture shows the iLBC codec measured measuring 4.21 narrowband focus scale.

 
For more information on making PESQ and POLQA measurements, ensure you contact only renown and well-respected test vendors because the science of speech quality measurements requires expertise and experience in many different areas audio, analog electronics as well as computing. It is easy to make a measurement but care is required to ensure that measurement is accurate and correlates to human subjective experience


The most trusted vendor for speech quality metrics is Malden Electronics, available in USA through Teraquant Corporation – www.teraquant.com


See use in http://technorati.com - X84QTD2E9BS6

Can You Hear Me Now?

PolQA is the new ITU-T Standard for Speech Quality Measurement which embraces Wideband or High Definition Telephony.

“Can you hear me now?”

We’ve all heard the refrain. How often have you been on a mobile phone & not been able to hear your calling party? How often have you experienced drop-outs on a VoIP call and missed that vital clue, that's important piece of information the caller mentioned which allowed you to understand their needs. May be you lost the business as a result. Good clear speech quality means productivity, both in business and in personal life. Everyone is critically busy these days and if you have to ask folks to repeat themselves, you waste time, first-rate meaningful conversation and miss information.

The existing telephony network uses 200-34000Hz analog bandwidth, digitized at a sampling rate of 8kbps. 8 bits of vertical resolution multiplied by 8kbps gives the traditional 64kbps bandwidth required for a voice channel. Compression by codecs such as G.729 and iLBC VoIP and specifically iSAC for Skype and GSM-FR & EVRC for wireless transmits narrowband traditional telephony at data rates as low as 4kbps.
So now we can compress voice sports to very low bandwidths and at the same time we have broadband Internet. so what can we do to improve speech quality.

Wideband or High Definition Telephony technology is now appearing in VoIP networks and wireless networks using voice codecs such as G.722 and WB-AMR. This provides speech with an analog bandwidth up to 7kHz and gives a richer listening experience. Those problems you currently have trying to recognize which of your young nieces or nephews is speaking to you is due to high frequencies filtered out with narrowband telephony. Wideband telephony will reinstate these, enriching your telephone conversation experience and improving productivity through speech clarity. This technology will eventually send telephony speech all the way up to 20 kHz, the limit of human hearing, equivalent to hi-fi music systems.

3gpp release 5 introduces AMR-WB codec which gives enhanced speech quality using data rates of only 16kbps. So wideband telephony or high definition telephony is being made available to wireless cellular networks.

Tools to Automatically Measure Speech Quality

Determining the subjective speech quality of a transmission system has always been an expensive and laborious process. The tool described in ITU-T Rec. P.862 Perceptual Evaluation of Speech Quality – PESQ provides a rapid and repeatable result in a few moments. PESQ is an objective measurement tool i.e. a computer measures the quality of the received audio in relation to the audio that was transmitted. PESQ predicts or has a very accurate close correlation to the results of subjective listening tests [i.e. human beings listening to speech files] On telephony systems. The resulting quality score is analogous to the subjective “Mean Opinion Score” (MOS) measured using panel tests according to ITU-T P.800. Strictly speaking, MOS is a score derived from human subjective testing. The PESQ scores are calibrated using a large database of subjective tests.

The ITU-T selection process that resulted in the standardization of PESQ involved a wide range of conditions, with demanding correlation requirements set to ensure that it has good performance in assessing conventional fixed and mobile networks and packet-based transmission systems.

Since ITU-T Rec. P.862 was originally released in 2000, further mappings of the PESQ score have been created. PESQ-LQ modified the score to improve correlation with subjective test results at the high and low ends of the scale where the raw PESQ score was found to be less accurate. A new mapping described in ITU-T Rec. P.862.1 was been released that further modified the raw score and correlated better to subjective testing.

PESQ Shortcomings - Time Warping

PESQ takes into account coding distortions, errors, packet loss, delay and variable delay, and filtering in analogue network components. The user interfaces have been designed to provide a simple access to this powerful algorithm, either directly from the analogue connection or from speech files recorded elsewhere.

PESQ Shortcomings
Noise Reduction:  (Subjective  >  PESQ)

The performance of a network or a network element can be fully characterized using high quality analog test equipment and PESQ. High quality analog interfaces are needed because the test equipment itself very easily introduces impairments which are included in the measurement and drank the desk score lower than should be measured for the system under test or network element. Whilst it is possible to use phonetically balanced sentences and other test patterns, accurate and repeatable measurements of the active speech level, activity, delay, echo, noise and speech quality can be obtained quickly using artificial speech test stimulus in different languages, which comprehensively tests all voice sounds the codec may be incident with, but at the same time achieves the process quickly in a time efficient way. A graphical mapping of the errors provides a useful insight into how the signal has been degraded and exactly what kind of sounds course the codec core system and test problems.

Since the launch of PESQ in 2000, there have been many advances in codec design. Unfortunately, PESQ was not trained on these later designs and can produce scores that are lower than expected from subjective tests. Time-warping and voice quality enhancement techniques are particularly difficult for PESQ. The ITU agreed on a new standard, P.863 POLQA, in 2010. POLQA addresses many of the issues and produces reliable scores for codecs, both old and new. POLQA is Now available on a couple of speech quality measurement platforms but Malden is the only platform that provides a quiet, high-quality analog front-end and the only platform to be recommended.