Is it possible to set ptime value when using the SIP Layer (Codec Opus ) ? on some few cases we have a robotic/chunky voice effet and we work on a stack using a ptime value of 20msec

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Since the problem is a bit random (but seems to appear recently), is there a way, on premise, to set packet time on Livekit Opus codec ?

Thanks — regarding setting Opus

ptime

on an on‑prem LiveKit SIP setup: I don’t have a direct configuration reference here. To determine if and how you can force/advertise a

ptime

(or negotiate

maxptime

) I need a few details: • LiveKit version and on‑prem deployment type (helm/k8s, binary, etc.). • Which SIP gateway/stack you use (kamailio/rtpengine, Asterisk, FreeSWITCH, PJSIP, etc.). • Example SDP from the SIP exchange (INVITE and 200 OK) or the portion showing

ptime

,

maxptime

, or

fmtp

lines for Opus. • Whether endpoints are WebRTC browsers or SIP endpoints and whether the issue is one‑way or both. • Whether you’ve tried setting

ptime

in the gateway or in SDP and where that change was applied. Share whichever you can (you can redact IPs/caller IDs). Once I have that I’ll tell you whether

ptime

can be set in your setup and the exact steps or alternatives to try (SDP munging, gateway config, or client changes). , with interactive elements