[Freeswitch-users] Strange and annoying problem.

[Carmelo Fischetti - Digi Unit]

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 at fscch00.dyndns-server.com entering state [calling][0]
2011-06-10 20:10:51.789283 [INFO] sofia.c:748
sofia/internal/1010 at 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 at 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 at 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 at fscch00.dyndns-server.com entering state [calling][0]
2011-06-10 20:32:32.965833 [INFO] sofia.c:748
sofia/internal/1010 at 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 at 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 at 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