# Interface RtcStats

WebRTC Stats of a Voice Call

interface RtcStats {
get audioRecvBytes(): any;
get audioRecvJitter(): any;
get audioRecvPackets(): any;
get audioRecvPacketsLost(): any;
get audioRtt(): any;
get audioSentBytes(): any;
get audioSentJitter(): any;
get audioSentPackets(): any;
get audioSentPacketsLost(): any;
}

#####
Index

### Accessors

audioRecvBytes
audioRecvJitter
audioRecvPackets
audioRecvPacketsLost
audioRtt
audioSentBytes
audioSentJitter
audioSentPackets
audioSentPacketsLost

## Accessors

### audioRecvBytes

- get audioRecvBytes(): any

- #### Returns any

### audioRecvJitter

- get audioRecvJitter(): any

- #### Returns any

### audioRecvPackets

- get audioRecvPackets(): any

- #### Returns any

### audioRecvPacketsLost

- get audioRecvPacketsLost(): any

- #### Returns any

### audioRtt

- get audioRtt(): any

- #### Returns any

### audioSentBytes

- get audioSentBytes(): any

- #### Returns any

### audioSentJitter

- get audioSentJitter(): any

- #### Returns any

### audioSentPackets

- get audioSentPackets(): any

- #### Returns any

### audioSentPacketsLost

- get audioSentPacketsLost(): any

- #### Returns any