[Freeswitch-users] Freeswitch seems to be giving conflicting timestamps - what's the deal here?

[Phil Quesinberry]

Looking into an intermittent problem that we're having with one of our DIDs,
I just noticed that some of the logged timestamps don't seem to make sense,
it's like there are two different clocks at work here.  What's interesting
is that I'm seeing this wrong time in received contact header info as well
but I'm guessing that's coming back to us from our own NTP timestamp in the
SDP we send back to them after the initial invite.  The system time is set
correctly.

Here is a short excerpt of the console log with siptrace enabled on the
external profile demonstrating this behavior:
recv 939 bytes from udp/[209.216.2.202]:5060 at 16:10:58.987966:
   ------------------------------------------------------------------------
   INVITE sip:1410xxx7612 at 173.59.xxx.xxx:5080;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 209.216.2.202:5060;branch=z9hG4bK68239f89;rport
   From: "+1410xxx3048"
<sip:+1410xxx3048 at 209.216.2.202>;tag=as0e92c0c82014-06-17 12:07:58.767052
[DEBUG] switch_core_session.c:1034 Send signal
sofia/external/+1410xxx3048 at 209.216.2.202 [BREAK]

   To: <sip:1410xxx7612 at voip.ourdomain.com:5080>;tag=Hj722eXX0yj8S
   Contact: <sip:+1410xxx3048 at 209.216.2.202>2014-06-17 12:07:58.767052
[DEBUG] switch_core_session.c:1034 Send signal
sofia/external/+1410xxx3048 at 209.216.2.202 [BREAK]

   Call-ID: 63cd75ae5caa77be67d5a6944e86f5fe at 209.216.2.202
   CSeq: 103 INVITE
   User-Agent: DidForSale-GW
   Max-Forwards: 70
   Remote-Party-ID: "+1410xxx3048"
<sip:+1410xxx3048 at 209.216.2.202>;privacy=off;screen=no
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
   Content-Type: application/sdp
   Content-Length: 289

   v=0
   o=root 2802 2803 IN IP4 207.223.69.137
   s=session
   c=IN IP4 207.223.69.137
   t=0 0
   m=audio 51480 RTP/AVP 0 8 18 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:18 G729/8000
   a=fmtp:18 annexb=no
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   ------------------------------------------------------------------------
send 405 bytes to udp/[209.216.2.202]:5060 at 16:10:58.988281:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 209.216.2.202:5060;branch=z9hG4bK68239f89;rport=5060
   From: "+1410xxx3048" <sip:+1410xxx3048 at 209.216.2.202>;tag=as0e92c0c8
   To: <sip:1410xxx7612 at voip.ourdomain.com:5080>;tag=Hj722eXX0yj8S
   Call-ID: 63cd75ae5caa77be67d5a6944e86f5fe at 209.216.2.202
   CSeq: 103 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.5.6b+git~20131022T212715Z~20a55d3cc6
   Content-Length: 0

   ------------------------------------------------------------------------

2014-06-17 12:07:58.787035 [DEBUG] sofia.c:5863 Channel
sofia/external/+1410xxx3048 at 209.216.2.202 entering state [ready][200]
2014-06-17 12:07:58.787035 [DEBUG] sofia.c:5863 Channel
sofia/external/+1410xxx3048 at 209.216.2.202 entering state [received][100]
2014-06-17 12:07:58.787035 [DEBUG] sofia.c:5873 Remote SDP:
v=0
o=root 2802 2803 IN IP4 207.223.69.137
s=session
c=IN IP4 207.223.69.137
t=0 0
m=audio 51480 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

2014-06-17 12:07:58.787035 [DEBUG] switch_core_media.c:2990 Audio Codec
Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2014-06-17 12:07:58.787035 [DEBUG] switch_core_media.c:1879 Already using
PCMU
2014-06-17 12:07:58.787035 [DEBUG] switch_core_media.c:3161 Set 2833 dtmf
send/recv payload to 101


Is this expected behavior or is this a bug in the version we're using?

Thanks.

Phil Quesinberry
Q Systems Engineering, Inc.
Embedded Hardware and Software Development
(410) 969-8002
http://www.qsystemsengineering.com <http://www.qsystemsengineering.com/> 


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