I am getting this issue recently. It is happening when answer call and no sound hear. I have customers so this is a big problem for us.
sip version: 1.1.0
livekit version 1.9.1

sip_server | github.com/livekit/sip/pkg/media/opus.(*encoder).WriteSample sip_server | /workspace/pkg/media/opus/opus.go:148 sip_server | github.com/livekit/media-sdk.(*SwitchWriter).WriteSample sip_server | /go/pkg/mod/github.com/livekit/[email protected]/media.go:150 sip_server | github.com/livekit/media-sdk.(*resampleWriter).flush sip_server | /go/pkg/mod/github.com/livekit/[email protected]/resample_soxr.go:131 sip_server | github.com/livekit/media-sdk.(*resampleWriter).WriteSample sip_server | /go/pkg/mod/github.com/livekit/[email protected]/resample_soxr.go:157 sip_server | github.com/livekit/media-sdk/g711.(*ULawDecoder).WriteSample sip_server | /go/pkg/mod/github.com/livekit/[email protected]/g711/ulaw.go:92 sip_server | github.com/livekit/media-sdk/rtp.(*MediaStreamIn[...]).HandleRTP sip_server | /go/pkg/mod/github.com/livekit/[email protected]/rtp/rtp.go:233 sip_server | github.com/livekit/sip/pkg/sip.(*rtpStatsHandler).HandleRTP sip_server | /workspace/pkg/sip/media.go:162 sip_server | github.com/livekit/sip/pkg/sip.(*rtpHandlerCount).HandleRTP sip_server | /workspace/pkg/sip/media.go:266 sip_server | github.com/livekit/media-sdk/rtp.(*Mux).HandleRTP sip_server | /go/pkg/mod/github.com/livekit/[email protected]/rtp/mux.go:81 sip_server | github.com/livekit/sip/pkg/sip.(*rtpHandlerCount).HandleRTP sip_server | /workspace/pkg/sip/media.go:266 sip_server | github.com/livekit/sip/pkg/sip.(*MediaPort).rtpReadLoop sip_server | /workspace/pkg/sip/media_port.go:455 sip_server | 2025-09-27T12:02:01.501Z DEBUG sip sip/media_port.go:466 handle RTP failed {"nodeID": "NE_KUqKqwrXjMHQ", "sipTrunk": "ST_wCbxhmaGWxka", "callID": "SCL_FvVfKSmnfFhE", "room": "68d7d22a3b83f622c63c9ee3", "participant": "phone_user", "participantName": "", "fromHost": "", "fromUser": "902162350973", "toHost": "sip.verimor.com.tr", "toUser": "905051319694", "ssrc": 1106677771, "payloadSize": 160, "rtpHeader": {"Version":2,"Padding":false,"Extension":false,"Marker":false,"PayloadType":0,"SequenceNumber":8484,"Timestamp":1106563952,"SSRC":1106677771,"CSRC":[],"ExtensionProfile":0,"Extensions":null,"PayloadOffset":0}, "pipeline": "Mux (T=0 -> StatsHandler(PCMU/8000) -> RTP(8000) -> PCMU(decode) -> Resample(8000->48000) -> Switch(48000) -> OPUS(encode) -> LKSamples(48000)) (T=101 -> StatsHandler(telephone-event/8000) -> HandlerFunc) (Def -> StatsHandler() -> HandlerFunc)", "errorCount": 3, "error": "opus: invalid argument"} sip_server | 2025-09-27T12:02:01.541Z