Hello, I'm trying to play from file when other party is set on-hold. reINVITE sent by FreeSwitch contains sendonly attribute which opens possibility that FreeSwitch can play something to other end although other party is on-hold. I've used modified switch_ivr_play_file() function which by default prevents sending rtp packets to other side when call is on-hold. I've encountered rtp stream issue causing distorted audio due to dropped packets on the caller end. Codec used is G.711 mulaw 8kHz and ptime is 20ms accepted on both side, FreeSwitch and softphone. Both endpoints are located in the local LAN. Wireshark rtp analysis shows that when call is not on-hold, packets comes in interval of 20ms as expected. But when the call is set on-hold, average delta time between packets is 21ms, causing dropping packet every second (1ms extra time multiplied by 50 packets per second) with jitter buffer set to 50ms in Wireshark. Setting "auto-jitterbuffer-msec" parameter in sofia did not help. Is there a serious reason why I cannot use switch_ivr_play_file() during on-hold and what could cause jitter only during on-hold period? When call is retrieved from hold, jitter disappears and audio is good again. Thanks, Miro