/ The VoIP druid
BuzzWhen working in the IP telephony world it is crucial to know how to use a network analyzer, in order to understand how the traffic is circulating over the network. This article presents some interesting Wireshark features related to voice over IP protocols.
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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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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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It can be useful or mandatory to save voice calls in order to be able to listen those afterwards. This is the often the case in call centres and in trading rooms, either for quality tracking or legal issues. Asterisk proposes for a long time now to save calls in sound files, this is called the monitoring.
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