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  • Even at the beginning of this decade, the above question might have appeared laughable, especially in the context of business organizations. Notwithstanding the transition to VoIP and the influx of mobile devices in the workplace, few people thought it would ever be possible to completely get rid of desk phones. Most assumed that the analog […]

    Can an Office do away with Deskphones?

    siponmobile.com
    Even at the beginning of this decade, the above question might have appeared laughable, especially in the context of business organizations. Notwithstanding the transition to VoIP and the influx of mobile devices in the workplace, few people thought it would ever be possible to completely get rid of desk phones. Most assumed that the analog […]

    Can an Office do away with Deskphones?

    siponiphone.com

    Reduce VoIP Latency and Echo by Manipulating the “PTime” Parameter

    Reducing the PTime Parameter with VoIP

    Reducing the PTime Parameter with VoIP

    Despite all the benefits that switching to a VoIP infrastructure can bring, there’s no doubt that it’s a more complicated system to set up. This is not because the PSTN system is inherently more or less easy to configure, but because it has had decades for telecommunication companies to standardize the various configuration aspects and to create uniform hardware to be used in our homes. VoIP being a new technology is still something that providers are experimenting with. Differing standards, various codecs, clock rates, NAT traversal protocols etc. can all contribute to configuration problems between differing devices and SIP providers. One issue that a lot of people have complained about is latency and echo with VoIP calls. There are certainly many factors causing both these issues such as the selection of an SIP provider who is a long distance away. But certain configuration tweaks on whatever system you use can help reduce these problems or eliminate them altogether. One such example is the parameters known as “ptime”.

    When converting a voice signal into a digital form to send using packets, the system needs to know how many seconds of voice data to include with each packet. This “packetization interval” is known as ptime and has a significant impact on various aspects of the VoIP set up. To start off with, a low ptime value implies a higher bandwidth requirement as well as a greater number of packets sent every second. For example, let’s say that 5 ms of voice data is encoded with each packet. This means that in the course of one second of talking, 400 individual packets need to be sent off. Different systems have different capabilities and so this packetization interval might be too less for some.

    The “standard” packetization interval or ptime parameter is 20 ms. It works quite well for a large number of systems. However, if you have the bandwidth to spare consider setting this to a lower value to reduce latency – the amount of time it takes for you to hear what the other person is saying. This makes sense because a large value implies that the system has to wait that much longer before it can send off a single packet. The more quickly it’s able to send the packets the quicker you can hear the other person. In addition to lowering latency, this can also reduce the amount of echo quite dramatically. The typical latency is equal to the round-trip time (rtt) + 2 * ptime so you can see that reducing the ptime has a pretty big impact on the total sum.

    Contact your SIP provider to find out whether or not you can reduce the packetization interval on your VoIP phones or whether  there are any disadvantages to doing so.

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