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1. Why do i need to set SNMP write
community on the Source Router ?
Both, the SNMP read and write community string needs to be set on the
source router. The write community is used to configure the IPSLA on
the device while the read community is used by OpManager to gather
performance data from the router. [back to top]
2. Why I am getting 'Source router SNMP write community may be
wrong' error message?
OpManager
uses SNMP to gather data from the Cisco IP SLA agent. This
error is displayed when wrong SNMP read / write community string is
configured for the Source router of the VoIP Monitor in
OpManager.
To
configure the correct SNMP write community string in OpManager, go to
the snapshot page of the source router and change the SNMP credentials
by clicking on the 'Click
here to change' corresponding to the "Passwords"
field. In the pop-up enter the appropriate credentials and submit it.
After successfully submitting the correct SNMP credentials, try to add
the VoIP Monitor again for the Source device (Maps > VoIP Monitor
> Settings). [back to top]
3. Why should the SLA
Responder be enabled
on the destination device ?
Enabling the IP SLAs Responder provides the details of packet loss
statistics on the device sending IP SLAs operations. IP SLAs Responder
is enabled on the target router (rtr responder) before configuring a
Jitter operation. [back to top]
4. Why are the VoIP
metrics shown as zero
or 'Not available' in OpManager?
You will see zero or 'not available' values when data is not collected
for the monitored metrics. This can be either due to incorrect SNMP
read community configured, or of the Responder is not enabled on the
destination device. Make sure that the correct SNMP read community is
configured and the SLA Responder is enabled. [back to top]
5.What are the critical
parameters monitored to determine the VoIP QoS performance?
The monitored parameters include Latency, Jitter, Packet Loss, and MOS.
The parameters are described below for reference:
Jitter :
Jitter is defined as a variation in the delay of received packets.
Users often experience disturbing sounds over a conversation coupled
with loss of synchronization at times and is referred to as jitter.
High levels of jitter can result in some packets getting discarded and
thereby impact the call quality. Ensuring a jitter-free transmission to
provide qualitative service depends on identifying the bottle-neck
responsible for the jitter, and acting on it to eliminate it.
OpManager's VoIP monitoring feature helps you find the problem and
ensures maximum QoS on your VoIP network.
Packet Loss : Packet
loss is a measure of the data lost during transmission from one
resource to another in a network. Packets are discarded often due to
network latency. Using OpManager, you can monitor the packet loss and
take corrective actions based on the information.
One way Latency:
Latency (delay) is the time taken for a packet to reach the destination
device. When monitoring latency over VoIP, the delay measured is the
time taken for a caller's voice at the source site to reach the other
caller at the destination site. Network latency contributes to delay in
voice transmission, resulting in huge gaps between the conversation and
interruptions.
Round Trip Time:
Round Trip Time is the time taken for a packet to reach the destination
and again comes back to the source device. The total time it takes for
the round trip is measured in milliseconds.
MOS: The
Mean Opinion Score is the key quality indicator of VoIP traffic
quality. And is measured in the scale of 1 to 5 (poor to excellent
quality). [back to top]
6. What is
VoIP codec?
Codecs
(Coder/Decoder) serve to encode voice/video data for transmission
across IP networks. The compression capability of a codec facilitates
saving network bandwidth and it is therefore appropriate that you
choose the correct codec for your IP network. Here is a quick reference
to the codecs with the corresponding packets size and bandwidth usage:
Codec & Bit Rate (Kbps) |
Operation Frequency |
Default number of packets |
Voice Payload Size |
Bandwidth |
Bandwidth |
Bandwidth |
G.711a/u |
60 msecs by default. You can specify in the range of 0 - 604800 msecs. |
1000 |
160 + 12 RTP bytes |
82.8 kbps |
67.6 |
87.2 |
G.729 |
1000 |
20 + 12 RTP bytes |
26.8 kbps |
11.6 |
31.2 |
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