NEC UNIVERGE SL2100, Networking Manual

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• Switching/Routing Delay
Switching/routing delay is the time the router takes to switch the packet. This time is needed to
analyze the packet header, check the routing table, and route the packet to the output port. This
delay depends on the architecture of the switches/routers and the size of the routing table.
• Queuing Time
Due to the statistical multiplexing nature of IP networks and to the asynchronous nature of packet
arrivals, some queuing, thus delay, is required at the input and output ports of a packet switch. This
delay is a function of the traffic load on a packet switch, the length of the packets and the statistical
distribution over the ports. Designing very large router and link capacities can reduce but not
completely eliminate this delay.
Jitter
Delay variation is the difference in delay exhibited by different packets that are part of the same traffic
flow. High frequency delay variation is known as jitter. Jitter is caused primarily by differences in queue
wait times for consecutive packets in a flow, and is the most significant issue for QoS. Certain traffic
types, especially real-time traffic such as voice, are very intolerant of jitter. Differences in packet arrival
times cause choppiness in the voice.
All transport systems exhibit some jitter. As long as jitter falls within defined tolerances, it does not
impact service quality. Excessive jitter can be overcome by buffering, but this increases delay, which
can cause other problems. With intelligent discard mechanisms, IP telephony/VoIP systems try to
synchronize a communication flow by selective packet discard, in an effort to avoid the walkie-talkie
phenomenon caused when two sides of a conversation have significant latency. NEC SL2100
incorporates a Jitter Buffer to avoid these problems.
Packet Loss
During a voice transmission, loss of multiple bits or packets of stream may cause an audible pop that
can become annoying to the user. In a data transmission, loss of a single bit or multiple packets of
information is almost never noticed by users. If packet drops become epidemic, the quality of all
transmissions degrades. Packet loss rate must be less than five percent for minimum quality and less
than one percent for toll quality.
2.2 Voice Quality Improvements
This section describes various techniques that can be used to improve the voice quality.
• Increase available bandwidth:
This can sometimes be the most basic solution and the easiest of the solutions. If running a System
IP Phone using G.711 with a 30ms fill time over Ethernet, for only one call, 90Kbps of bandwidth is
needed. If that same user only has a 64K line, they do not have a decent IP voice call. The user can
increase the available bandwidth to slightly exceed the 90Kbps requirements and their voice quality
dramatically increases. This solution might not be viable if no more bandwidth is available.
• Use a different CODEC:
The CODEC contains possible compression algorithms to be used on the voice. Let’s take the
example above again. The user only wants one voice line over a 64Kbps data connection. They
also want to maintain their current fill time of 30ms. Change to a G.729. For one line, only 34Kbps is
required for a call. This fits well within the 64Kbps of available bandwidth.
• Increase the number of frames per packet:
To continue with the example above, the user has moved to a G.729 CODEC. But now, the user
wishes to add two more System IP Phones. Their current 64Kbps line can handle one call, because
it is only 34Kbps. Two more System IP Phones would increase the total to 102Kbps so obviously
there is not sufficient bandwidth.
The user can now increase the fill time to 50ms. This reduces the bandwidth per call to 19.8Kbps
(3x 19.8 = 59.4Kbps). The savings in bandwidth comes from the fact that with a longer fill time,
fewer packets are needed to send the voice. With fewer packets, less header information needs to
be attached and transmitted.
SL2100 ISSUE 1.0
6-2 Network Design Considerations