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Old 18-03-2011, 05:30 PM
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M.Arsalan Qureshi

 

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Default Packet Loss and Echo on VoIP Networks


Packet Loss and Echo on VoIP Networks

Packet loss can occur on any network for a variety of reasons including congested links (packets dropped when buffers are full), routing problems, incorrect equipment configuration issues, and more. Because voice traffic uses UDP as its transport protocol, dropped packets are lost, and obviously not resent. On a voice conversation, this is recognized by what appears to be speech that is cut short or clipped – portions of the voice conversation might be lost, where one user speaking “Hello, may I speak to Dan?” might sound more like “Hello, may… Dan?”, or similar.


The codecs outlined in previous articles are typically capable of correcting (via a DSP) for up to 30 ms of lost voice traffic without any noticeable impact on quality. However, Cisco voice packets use a 20 ms payload, so effectively only one packet could be lost at any point in time. In order to avoid packet loss issues, it is important that the underlying IP network is properly designed (including redundancy), and that QoS techniques are implemented effectively.


Another issue that impacts voice calls is one that you have likely already experienced, namely the sound of your own voice echoing back to you a short time after speaking. Echo occurs when part of the voice transmitted “leaks” back on to the return path of a call. To compensate for this, most codecs use built-in echo cancellation techniques. On a Cisco gateway (such as a router), echo cancellation settings are configured by default, but can be tuned in order to compensate for different degrees of echo experienced by users.
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