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The two fundamental duties of a softPBX are call signaling and voice transmission
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Voice channels in a VoIP network are analogous to loops or circuits in a traditional voice network
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Voice transmission occurs across virtual channels on an IP network, most frequently using UDP packets because they have less overhead than TCP packets
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Codecs are algorithms that digitize and package sound and/or video for transport across the network.
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The most common codec for LAN use is G.711
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G.711 uses one of two digitizing scales : m law or Alaw. m law is the standard in North America. Each requires 64 kbps of bandwidth
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As a rule, TCP/IP and Ethernet add 32 kbps overhead to the bandwidth required by the codec
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Selection of the codec for each call is accomplished by the signaling function of the softPBX or by direct negotiation between two endpoints at the beginning of that call
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Transcoding is the real-time translation of one codec or another. This may be necessary when two endpoints cannot use the same codec or when a conferencing application dictates the use of more than one codec to support multiple endpoints.
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Different codecs offer different levels of bandwidth economy.
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The top causes of perceived quality problems in voice transmission are jitter, lag, and packet loss. Generally, the more bandwidth-conservative a codec is, the more prone to noticeable quality breakdowns caused by these issues
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A call path is the chain of virtualized voice channels used to connect a call across the network. Each channel may employ a different codec
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A call path that is established directly between two endpoints, without a softPBX, proxy, or transcoding element in between, is an independent call path.
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Independent call paths result in lower processing intensity because the softPBX doesn't have to handle the audio. Complex, highly compressed codecs and softPBX call paths result in higher processing intensity