Hi, all We have some unclear situation. Call flow is below. 1. A --INVITE(SDP)--> FS C 2. A FS --INVITE(SDP)--> C 3. A FS <--183(SDP)-- C 4. A <--183(SDP) FS C 5. A FS <--RTP-- C 6. A <--RTP-- FS C 7. A <--480-- FS C 8. A FS --CANCEL--> C FreeSwitch (FS) sends to 'A' SIP 480 (Reason header: SIP;cause=604;text="MEDIA_TIMEOUT"). There is no RTP from 'A' while progress. Time between 4. and 7. - 30 seconds. Is this expected behaviour or we should update? Current version - FreeSWITCH Version 1.5.6b+git~20131029T184027Z~88a13f1fc2~32bit (git 88a13f1 2013-10-29 18:40:27Z 32bit)