![]() ![]() You should consider that in an SfB environment you’ll be using different codecs in different scenarios. ![]() Latency is measured as one-way or Round-trip Time (RTT).ĭifferent codecs deal with imperfect networks better or worse, modern codecs like RTAudio and Silk dealing better with network issues than older codecs like G711. This network propagation delay is essentially tied to the physical distance between the two points and the speed of light, including additional overhead taken by the various routers in between. Latency This is the time it takes to get an IP packet from point A to point B on the network.It’s only when the jitter exceeds the buffering that a participant will notice the effects of jitter. Most modern VoIP software including Skype for Business can adapt to some levels of jitter through buffering. Inter-packet arrival jitter or simply jitter This is the average change in delay between successive packets. ![]() Packet loss directly affects audio quality-from small, individual lost packets having almost no impact, to back-to-back burst losses that cause complete audio cut-out.
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