carriers-sip
Packet loss
Packet loss happens when voice data packets fail to reach their destination, leaving gaps that sound like clipped words or robotic, broken audio.
Packet loss is what happens when small chunks of your call audio leave one side but never arrive at the other. Voice travels over the internet as a stream of tiny packets carried by RTP (Real-time Transport Protocol). If some of those packets get dropped along the way, the receiving end is missing pieces of the conversation, and your agents and leads hear clipped words, dead spots, or a robotic warble.
A little loss is normal on any network, but it adds up fast on a busy dialer. Because Predictive dialing can push many Concurrent calls at once, your connection has to carry a lot of audio at the same time. When the link is too full, packets get discarded, and call quality drops for everyone. Loss above roughly one to two percent is usually noticeable.
Common causes
- A saturated internet link, where total call traffic exceeds available bandwidth.
- Cheap or overloaded network gear that drops packets under load.
- Wi-Fi for agents instead of a wired connection.
Packet loss is closely tied to Jitter and Latency, and together they drag down your MOS (mean opinion score) score, which is the standard way to grade call quality. If you only hear problems in one direction, that is more likely a One-way audio issue than plain loss. To diagnose, count dropped packets at the network level rather than guessing from how a call sounds. Picking an efficient Codec such as G.729 codec uses less bandwidth and gives each packet more room to arrive intact.
The frustrating thing about packet loss is that it is intermittent. A call that sounds fine at ten in the morning can fall apart at two in the afternoon when your team is dialing flat out. That is why a one-off speed test rarely tells you anything useful. You want to watch loss over time, ideally during your busiest hours, so you can see whether it climbs as call volume rises. If it does, the link or the gear is the bottleneck, not the carrier. Fix the network first, because no amount of carrier tuning will help if your own connection is dropping the audio before it ever leaves the building.
Related terms
Codec
The method that compresses and decompresses voice audio for a VoIP call, trading off between sound quality and how much network bandwidth each call uses.
Jitter
Jitter is the uneven arrival timing of voice packets on a call, which can make audio sound choppy or robotic even when little data is actually lost.
Latency
Latency is the delay between speaking and being heard on a call — the time audio takes to travel across the network, measured in milliseconds.
MOS (mean opinion score)
MOS, or mean opinion score, is a one-to-five rating of call audio quality, where five is perfect and anything below about three sounds noticeably bad.
One-way audio
One-way audio is when one person on a call can be heard but cannot hear the other, almost always a problem with how the audio stream is routed.
RTP
Real-time Transport Protocol, the stream of small packets that carries the actual voice audio of a VoIP call between the two endpoints.