Hello to everybody, I'm new to this list but I've been experimenting with FreeSWITCH for 3 months. My environment consists of a FreeSWITCH server and some (4-5) SIP softphones. We are noting a strange and annoying problem. When in call, after exactly 15 minutes, FreeSWITCH CLI logs the following lines: 2011-06-10 20:10:51.689289 [DEBUG] sofia.c:4769 Channel sofia/internal/1010@fscch00.dyndns-server.com entering state [calling][0] 2011-06-10 20:10:51.789283 [INFO] sofia.c:748 sofia/internal/1010@fscch00.dyndns-server.com Update Callee ID to "Outbound Call" <1010> 2011-06-10 20:10:51.789283 [DEBUG] sofia.c:4769 Channel sofia/internal/1010@fscch00.dyndns-server.com entering state [ready][200] 2011-06-10 20:10:51.789283 [DEBUG] sofia.c:4780 Remote SDP: v=0 o=amsip 0 0 IN IP4 192.168.1.100 s=talk c=IN IP4 192.168.1.100 t=0 0 m=audio 12100 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=setup:both a=ptime:20 2011-06-10 20:10:51.789283 [DEBUG] sofia_glue.c:4658 Audio Codec Compare [GSM:3:8000:20:13200]/[GSM:3:8000:20:13200] 2011-06-10 20:10:51.789283 [DEBUG] sofia_glue.c:2724 Already using GSM 2011-06-10 20:10:51.789283 [DEBUG] sofia_glue.c:4772 Set 2833 dtmf send/recv payload to 101 2011-06-10 20:10:51.789283 [DEBUG] sofia.c:5238 Processing updated SDP 2011-06-10 20:10:51.789283 [DEBUG] sofia_glue.c:3009 Audio params are unchanged for sofia/internal/1010@fscch00.dyndns-server.com. My first question is: what does this mean? Is there some configuration parameter (where?) that triggers some event after 15 minutes? Actually, for unencrypted calls, nothing happens but when using SRTP, the log changes to: 2011-06-10 20:32:32.825843 [DEBUG] sofia.c:4769 Channel sofia/internal/1010@fscch00.dyndns-server.com entering state [calling][0] 2011-06-10 20:32:32.965833 [INFO] sofia.c:748 sofia/internal/1010@fscch00.dyndns-server.com Update Callee ID to "Outbound Call" <1010> 2011-06-10 20:32:32.965833 [DEBUG] sofia.c:4769 Channel sofia/internal/1010@fscch00.dyndns-server.com entering state [ready][200] 2011-06-10 20:32:32.965833 [DEBUG] sofia.c:4780 Remote SDP: v=0 o=amsip 0 0 IN IP4 192.168.1.100 s=talk c=IN IP4 192.168.1.100 t=0 0 m=audio 12665 RTP/SAVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=setup:both a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:YjUwMzQyNWU4YmY2ZjhiOGYyYTQ1MmZiOGNhMWEy a=encryption:optional a=ptime:20 2011-06-10 20:32:32.965833 [DEBUG] sofia_glue.c:4463 Existing key is still valid. 2011-06-10 20:32:32.965833 [DEBUG] sofia_glue.c:4658 Audio Codec Compare [GSM:3:8000:20:13200]/[GSM:3:8000:20:13200] 2011-06-10 20:32:32.965833 [DEBUG] sofia_glue.c:2724 Already using GSM 2011-06-10 20:32:32.965833 [DEBUG] sofia_glue.c:4772 Set 2833 dtmf send/recv payload to 101 2011-06-10 20:32:32.965833 [DEBUG] sofia.c:5238 Processing updated SDP 2011-06-10 20:32:32.965833 [DEBUG] sofia_glue.c:3009 Audio params are unchanged for sofia/internal/1010@fscch00.dyndns-server.com. 2011-06-10 20:32:33.485800 [ERR] switch_rtp.c:2560 Error: SRTP unprotect failed with code 7 (auth check failed) (the last line repeated indefinitely)... and one leg of the call is dropped (one of the phones cannot hear the other). Since this happens with various softphones, it's a server issue, I suppose. Any help would be greatly appreciated. Thanks in advance. Carmelo