# Client-side call quality

DevFeed: [Client-side call quality](<https://devfeed.tech/articles/client-side-call-quality-1748.md>)

Original publisher: [Read original article](<https://signal.org/blog/client-side-audio-quality/>)

Published: 2013-02-18T00:00:00Z

Content type: article

Language: en

Sources: [Signal Blog](<https://devfeed.tech/sources/signal-blog.md>)

Topics: [client](<https://devfeed.tech/topics/client.md>)

Tags: [audio](<https://devfeed.tech/tags/audio.md>), [latency](<https://devfeed.tech/tags/latency.md>), [mobile](<https://devfeed.tech/tags/mobile.md>), [networks](<https://devfeed.tech/tags/networks.md>), [real-time](<https://devfeed.tech/tags/real-time.md>), [streams](<https://devfeed.tech/tags/streams.md>)

## AI overview

The article explains how a mobile VoIP client uses an adaptive jitter buffer to balance audio latency against late or missing UDP audio packets under variable network conditions.

## Source excerpt

In our previous post, we discussed the global infrastructure that allows RedPhone clients to find low-latency servers when establishing a call. This post discusses the techniques we use to retain call quality when network conditions are less than ideal. Read more...