This document helps you understand and troubleshoot some voice quality
issues between two Cisco IP phones. In particular, this document addresses:
Cisco recommends that you have knowledge of this topic:
The information in this document is based on these software and
The information in this document was created from the devices in a
specific lab environment. All of the devices used in this document started with
a cleared (default) configuration. If your network is live, make sure that you
understand the potential impact of any command.
Technical Tips Conventions for more information on document
Complete these steps in order to troubleshoot echo on a Cisco IP
Use the Volume buttons to adjust the volume to
less than 75 percent on the calling and the called Cisco IP phones.
Press Settings on the IP
Select the Save softkey.
The two components that affect echo are amplitude (loudness of the
echo) and delay (the time between the spoken value and the echoed sound). You
can use suppressors or cancellers to control echo.
The two sources of echo are Hybrid echo and Acoustic echo.
Hybrid echo is caused by an impedance mismatch in the hybrid circuit,
such as a two-wire to four-wire interface. This mismatch causes the transmit
(Tx) signal to appear on the receive (Rx) signal.
Acoustic echo is caused by poor acoustic isolation between the earpiece
and the microphone in handsets and hands-free devices.
Echo is perceived as annoying when all of these conditions are true:
Signal leakage between analog Tx and Rx paths
Sufficient delay in echo return
Sufficient echo amplitude
The echo is independent of the far-end phone volume level and depends
only on the local phone level. In the past, side-tones were proportional to
volume, this effectively masks the problem. This solution is not possible here
because side-tones become too loud and the masks do not work as soon as the
jitter buffer depth increases.
It is thought that there is an antenna effect in the cord.
For additional troubleshooting information, locate the echo and
suppress it. Refer to
Analysis for Voice over IP for assistance.
In order to troubleshoot voice chopping or breaking, try to disable
Voice Activity Detection (VAD).
Choose Service > Service
Parameters on the Cisco CallManager.
Choose the server, and then choose Cisco Call
Under Clusterwide Parameters (Device-General), set Silence
Suppression and Silence Suppression for Gateways to
This disables Silence Suppression on all devices. In other words,
it turns VAD off.
Note: The Strip G.729 Annex B (Silence Suppression) from Capabilities
service parameter works independently from the other system-wide parameters
Silence Suppression and Silence Suppression for Gateways. Those parameters only
set the boolean silence suppression in Skinny (SCCP) and MGCP, but does not
affect the capabilities negotiation. Some devices do override the value of this
boolean based on the negotiated capability.
This problem can appear due to functionality of silence suppression/VAD
The Pulse Code Modulation (PCM) signal first passes through a
high-pass filter. The energy of each sample is measured, and the average speech
power is calculated. If the power is greater than -31 dBm0 (dBm0 equals a
digital milliwatt), the signal is recognized as speech.
When voice activity resumes after a period of silence (power less then
-31 dBm0), a certain period of time is required in order to determine that
voice resumes until silence suppression is turned off. During this period
between when voice activity starts and silence suppression ends, sent packets
are lost. Although the loss is only brief, the result is a noticeable
degradation of quality of voice to the end user. This is observed more when
there are short periods of silence between words (speech detector switches on
and off quickly).
If you require further troubleshooting, check for possible packet drops
in the switching network. Check for buffer overflows and queuing and refer to
the Cisco IP Telephony
Quality of Service (QoS) Design Guide for further information.
Choppy voice is experienced for callers who use the G.729 codec in a
remote site when they speak with external parties through the public switched
telephone network (PSTN). Internal IP phone-to-IP phone connections and those
that use the G.711 codec do not experience this problem, and the WAN bandwidth
is enough for the voice calls in a remote site.
This problem can arise because the G.729 codec that is used by the
remote site IP phone is G.792br8, which has Voice Activity Detection (VAD)
functionality built into the codec. One of the frequent causes of jitter and
choppy voice is VAD, as explained in the previous section.
In order to overcome this problem , set the Strip G.729 Annex B
(Silence Suppression) from Capabilities parameter to
True from the Cisco CallManager service parameter page. When
this parameter is set to True, this codec is eliminated from
the codecs available during codec negotiation. With this change implemented,
Cisco CallManager can not enable VAD.
The G.729 codec can have multiple variances, which are termed as
Annexes. AnnexB is G.729br8. With this codec,
VAD can not be disabled. The Cisco CallManager always tries to negotiate the
G.729br8 codec first. When you modify the Cisco CallManager service parameter,
you instruct the Cisco CallManager not to use G.729br8 and instead use the
G.729r8 codec, without any Annexes. The other option is to use a completely
different codec, such as G.711ulaw.
Press I on the Cisco 79xx twice during the call in
order to display information on the call in progress. Check these items:
Variance in IP transport time from packet to packet causes jitter.
Check that the size of the buffer and the difference between the
average jitter value and maximum jitter value are not too high. If they are,
there is a chance of a problem. This is because the swing is high and the
First-In, First-Out (FIFO) might not have enough time to compensate. If the
packets are stuck in FIFO, the delay of the stream causes the delay effect
known as long distance calls. If packets arrive too late or too early to be
useful and the jitter management mechanisms are unable to sort the arriving
packets into their original order, the voice play out is distorted and
Jitter is defined as a variation in the delay of received packets. At
the sending side, packets are sent in a continuous stream with the packets
spaced evenly apart. Due to network congestion, improper queuing, or
configuration errors, this steady stream can become lumpy, or the delay between
each packet can vary instead of remaining constant. There are many causes of
jitter behavior of voice, but common ones are congested networks, number of
hops over which voice travels or behavior of some queuing algorithms. Refer to
Jitter in Packet Voice Networks for more information.
Refer to the Cisco IP Telephony QoS Design
Guide for queuing problems.
QoS for IP Telephony for an understanding QoS for IP Telephony.
You experience a low volume on the IP phone even though the volume is
increased to maximum through the volume button.
This issue is occurs due to low gain of a signal that comes into the
voice port of router. If the voice level is too low, you can increase the input
gain with the input gain decibels command. Complete
these steps in order to resolve this issue:
Issue this command in the voice port:
Note: Gain, in decibels, to be inserted at the receiver side of the
interface. Range is integers from -6 to 14. The default is 0.
The shut and no
shut commands are required in order to accept the voice port
Change values, place test calls, and adjust the signal as needed.
Increase the input gain level by 1dB after each test call until optimal volume
IOS Voice Command Reference for more information.