User`s guide

Latency

is de ned as the average “travel” time it takes for a packet to pass through the network, from

source to destination. The average time varies according to the amount of traf c being transmitted and

the bandwidth available at that given moment. If the traf c is greater than the bandwidth available,

packet delivery will be delayed.

MultiVOIP deals with the latency issue in a private network as well as over the public Internet. In a

private network, when network traf c is at peak levels, voice can be given priority over data to ensure

consistently high voice quality using the Differentiated Services (DiffServ) Quality of Service (QoS)

protocol. This is an end-to-end requirement, which means it must be supported at various points on

the network in order for the voice traf c to receive the proper priority from every device it encounters.

Another way to enforce Quality of Service is to use the Resource Reservation Setup Protocol (RSVP).

RSVP-enabled routers set aside bandwidth along the route from source to destination based on the IP

addresses associated to the MultiVOIP gateways.

When running Voice over IP on the public

Internet, the issue of latency cannot be

controlled due to the ever-changing path and

router hops that your voice packet may take

before it reaches its destination. However,

the MultiVOIP gateway does a good job of not

adding any additional latency through the box

itself. Therefore, if you have a good Internet service provider, and they are able to provide you with a

quality of service guarantee, you should be able to manage any latency you may encounter.

If you have concerns about latency on your network, or the public Internet, use the above threshold

chart to determine its possible affect on your voice quality.

Jitter

is de ned as the variability in packet arrival at the destination. Voice packets must compete

with non real-time data traf c, therefore, if there are bursts of traf c on the network, they can result

in varied arrival times. When consecutive voice packets arrive at irregular intervals, the result is a

distortion in the sound, which if severe, can make the speaker unintelligible.

The MultiVOIP gateway utilizes a Dynamic Jitter Buffer to collect voice packets from the IP network,

store them, and shift them to the voice processor in evenly spaced intervals. During high latency

periods, the jitter buffer size is dynamically increased to receive delayed voice packets. During low

latency periods, the jitter buffer is dynamically decreased to minimize the end-to-end voice delay.

Packet loss

is the percentage of undelivered packets in the data network. When data packets are lost, a

receiving computer can simply request a retransmission. When voice packets are lost, or arrive too late,

they are discarded instead of retransmitted. The result is disconcerning gaps in the conversation (like a

poor cell phone conversation).

The MultiVOIP gateway utilizes Forward Error Correction to increase voice quality by recovering lost

or corrupted packets. The current Forward Error Correction implementation can recover one of two

consecutive lost/corrupted packets or every other lost/corrupted packet, thereby eliminating any

noticeable voice degradation.

Optimum Latency Thresholds and Voice Quality:

Up to 150 ms = excellent

150 - 250 ms = good

250 - 350 ms = usually acceptable

> 350 ms = depends on application