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VoIP Troubleshooting 101 – Packet Loss & Reliability April 6, 2010

Posted by TelUS Consulting Services in VoIP catagory.
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Packet Loss

Packet loss occurs for many reasons, and in some cases, is unavoidable. Often the amount of traffic a network is going to transport is underestimated. During network congestion, routers and switches can overflow their queue buffers and be forced to discard packets. Packet loss for non-real-time applications, such as Web browsers and file transfers, is undesirable, but not critical. The protocols used by non-real-time applications, usually TCP, are tolerant to some amount of packet loss because of their
retransmission capabilities.

Real-time applications based on UDP are significantly less tolerant to packet loss. UDP does not have retransmission facilities, however, retransmissions would almost never help. In an RTP session, by the time a media gateway could receive a retransmission, it would no longer be relative to the reconstructed voice waveform; that part of the waveform in the retransmitted packet would arrive too late.

It is important that bearer and signaling packets are not discarded, otherwise, voice quality or service disruptions might occur. In such instances, the Class of Service (CoS), which is equivalent to DSCP in IP, but at the Ethernet layer, mechanisms become very important. By configuring CoS parameters, administrators can give packets of greater importance a higher priority in the network, thus ensuring packet delivery for critical applications, even during times of network congestion.

Although packet loss of any kind is undesirable, some loss can be tolerated. Some amount of packet loss for voice services could be acceptable, as long as the loss is spread out over a large amount of users. As long as the amount of packet loss is less than five percent for the total number of calls, the quality generally is not adversely affected. It is best to drop a packet, versus increasing the latency of all delivered packets by further buffering them.

Reliability

Although network failures are rare, planning for them is essential. Failover strategies are desirable for cases when network devices malfunction or links are broken. An important strategy is to deploy redundant links between network devices and/or to deploy redundant equipment. To ensure continued service, organizations should plan carefully for how media gateways and media gateway controllers can make use of the redundant schemes.

IP networks use routing protocols to exchange routing information. As part of their operation, routing protocols monitor the status of interconnecting links. Routing protocols typically detect and reroute packets around a failure if an alternate path exists. Depending on the interconnecting media used for these links, the time taken to detect and recalculate an alternate path can vary. For example, the loss of signal for a SONET/SDH connection can be detected and subsequently rerouted very quickly. However, a
connection through an intervening LAN switch might need to time out the keep-alive protocol before a failure is detected.

Having media gateways and media gateway controllers that can actively detect the status of their next-hop address (default gateway) as part of their failover mechanism decreases the likelihood of a large service disruption. Another possible option is that the media gateway and media gateway controller could be directly connected to the router. In this case, the possibility of a link failure (depending on the nature of the failure) could be immediately detected and the network devices would take appropriate action. Still another option for reducing long-term failure could be to employ a redundancy mechanism such as the Virtual Router Redundancy Protocol (VRRP).

I hope someone found this series beneficial. Suggestions for the next series would be appreciated.

Joe Buck, N.C.E.

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