You can argue over exactly what level of the above metrics is “good” or “bad” depending on the codec and the tolerance of the user, ultimately there are no definitively correct or incorrect thresholds as long as they are used to find issues and improve network performance. 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.
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.