/ The VoIP druid
BuzzVoice transport should always balance between real time and bandwidth consumption. When interconnecting two PBX in order to exchange more than one communication, it could be interesting to limit bandwidth consumption by carrying more voice samples in the same IP frame. This is what the Asterisk Inter Exchange protocol (aka IAX) is doing and gains are real.
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Transporting voice over a packet network is not an easy task and requires controlled quality with regards to bandwidth first but also with delay, jitter and frame loss ratio. We can encounter a very bad voice quality during network congestion, which is not good for a professional network. In order to test our network in advance or during production there are tools and techniques that can be easily used. This article shows how to simply test an IP network for voice transport with off the shelf tools.
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As soon as we install in an organisation an Asterisk IP telephony service, it can impact directly on the service and we should study how it can be resilient. This article proposes a very simple but redundant telephony service with two Asterisk servers installed in a load-balancing mode.
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Transit delay on the IP network is a real issue when pushing voice communication on top of it, it really affects quality and comfort of users. We generally consider that a delay above 150 ms becomes sensible to human ear and above 400 ms is considered painful1 and forces users to apply delays before starting a new sentence. With Asterisk, it is possible to verify if the round-trip delay is compatible with the service quality we would like to achieve on our telephony over IP network: it is called qualification.
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The Counterpath new release of the X-lite softphone1 is now supporting the Speex codec. This one is a high definition codec, that is mainly called HD and promises an enhanced quality than G.711 standard PCM that we have on standard telephony. The main reason why is the encoding combined to a sampling performed on a wider frequency band (up to 32 kHz versus 8 kHz). This should push competition in the softphone and standard IP phone industry to move towards these new HD codecs and especially Speex which is available under an open source license. Further more, G.722 is not widely adopted at this time even with new high range product announced, like at Polycom on the 6xx serie.
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