Troubleshooting guide

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Cisco Broadband Local Integrated Services Solution Troubleshooting Guide
OL-5169-01
Chapter 4 Troubleshooting with Call Flows
Voice Quality Problems
Lost or Distorted Audio
One of the most common problems is a breaking up of audio, which is often described as garbled speech
or a loss of syllables within words or sentences. There are two common causes: packet loss and jitter.
• Packet loss—occurs when packets are dropped or arrive at their destination too late to be useful.
• Jitter—is the variation in arrival times of successive packets.
Ideally, all VoIP packets from one phone to another would arrive at a constant rate of 1 every 20 ms.
Notice that travel time itself does not matter, only the variation in arrival times between packets.
There are many sources of variable delay in a packetized cable network, and it cannot be eliminated
entirely. In anticipation of variable delay, digital signal processors (DSPs) on phones and other voice
capable devices are designed to buffer some of the audio.
This dejittering is done only after the audio packet has reached its destination and is ready to be put into
an audio stream (that is, played into the user’s ear or sent to the PSTN via a digital PCM stream). The
network administrator can minimize the variation in packet arrival times by applying quality-of-service
(QoS) and other measures in advance.
When faced with lost or distorted audio, first try to determine the path of the audio. Identify each
network device in the path of the call’s audio stream. Try to determine whether the problem occurs only
between two particular sites, only through a certain gateway, and so on. This will help to isolate the
devices to examine.
For troubleshooting, it may be desirable to disable silence suppression (also known as Voice Activation
Detection or VAD). This mechanism saves bandwidth by not transmitting audio when there is silence,
but may cause noticeable (and unacceptable) clipping at the beginning of words. You can then use a
network (or protocol) analyzer to view packet flow.
A call between two phones should have 50 packets per second (or 1 packet every 20 ms) with silence
suppression disabled. In a poor quality call (such as a call with a lot of jitter), the difference in arrival
times of these packets can vary greatly. With proper filtering, it should be possible to determine whether
packets are being lost or delayed excessively.
Crackling
Another symptom of poor voice quality may be crackling, which is sometimes caused by a defective
power supply or strong electrical interference close to the phone. Try swapping the power supply and
moving the phone and/or MTA to a different location.
Echo
Echo (also known as talker echo) occurs when a talker’s transmitted speech is coupled into the receive
path from the far end. Talkers then hear their own voices, delayed by the total echo path delay time. The
cause of echo usually lies in analog components and wiring. Exceptions occur when one party is using
a speakerphone whose volume is set too high, or in other situations that can create an audio loop.
When troubleshooting echo problems, make sure the phones being tested are not using a speakerphone
and they have the headset volume set to reasonable levels (start with 50 percent of the maximum audio
level). Most of the time, echo problems occur when attaching to the PSTN through a gateway.
Although the source of the problem is almost always at the far end, it is difficult to change anything in
the PSTN. So the first step is to determine which gateway is being used. It might be possible to add
additional padding to the gateway in the transmit direction (toward the PSTN) so that the lower signal
strength yields less reflected energy.