Author Topic: Stream Manager/SIP Server/CGSIPListener[13]: Exception on listen socket  (Read 4375 times)

Lumpy

  • Guest
Well, I had some audio files playing at one point then I had to start playing with things and yada, yada, yada; they don't play anymore.  Basically, I have a very simple strategy loaded on a route point that plays a front-end greeting (70000) and then plays musiconhold if the call waits in queue.  It looks like the CGSIPListener[13]: Exception on listen socket is the problem or the .wav file is incomplete, but I don't have any idea what that means.  Has anyone seen this before?


SIP Log snip:
To: <sip:annc@LHGPTLDEVGENSMTLX01:5070;play=announcement/70000;repeat=1>;tag=0093B4EE-1BA6-1769-9D3A-F900010AAA77-49
Call-ID: 008C88CC-2427-1769-9D01-0F00010AAA77-55@10.1.0.15
CSeq: 2 BYE
Content-Length: 0
Via: SIP/2.0/UDP 10.1.0.15:5060;branch=z9hG4bK008C89D0-2427-1769-9D01-0F00010AAA77-257


sipcs: CGSIPListener[13]: Exception on listen socket
sipcs: SIPTS_TreatmentDlg[66]: << EVENT(53) <<
sipcs: SIPTS_TreatmentDlg[66]: << EVENT(58) <<
sipcs: SIPTS_TreatmentDlg[66]: << DESTROY <<
sipcs: ~SIPTS_TreatmentDlg[0x9062410]


Stream manager log snip:

SIPdialog[51] event 12 INVITE
@10:36:48.7511 [53050] SIPdialog[51] annc@..;play=announcement/70000;repeat=1
  m=audio RTPport=19218 a=sendrecv
    pt=0 codec=3(G.711/mu-law)
    pt=3 codec=10(GSM)
    pt=8 codec=4(G.711/A-law)
    pt=101 telephone-event
Probing announcement/70000 (codecs=ua..g!...,fmt=wav)
  announcement/70000_mulaw.wav Ok
  announcement/70000_{pcma,alaw}.wav not found
  announcement/70000_{gsm,gsmFR}.wav not found
AnswerSDP(base=ua..g!...,mask=u....!...:ua..gm...,play)
  audio:0 codec=3(G.711/mu-law) nte:101 a=sendrecv
GKconnSIP::setupLeg(audio)
  RTPLeg[514] created RTP:8100(fd=99), RTCP:8101(fd=100)
RTPleg::setup [SM-7.5.005.03]-(rtp:8100)--(10.1.0.156:19218)
  (leg=514) state=67f[.rs.:ci:trs] codec=3(G.711/mu-law)
SIPconnAnnc::setup_media(leg=514,audio)
  RTPfile[3bc5d753] seq=28578 time=798799049
add_media_source(announcement/70000,wav,ua..gm...)
  WAVfile(announcement/70000_mulaw.wav) fmt=7(G.711/mu-law) chan=1 sample=8000/sec blk=1
    data_size=148480 (~18.6 sec)
  CachedFile(announcement/70000_mulaw.wav) size=148480

gsip:CL2CONN[12,UDP]:10:36:48.762 >>>> 696 bytes to 10.1.0.15:5060 >>>>
SIP/2.0 200 OK
From: "207xxxxxxx" <sip:207xxxxxxx@10.1.0.156>;tag=as1499e728
To: <sip:annc@LHGPTLDEVGENSMTLX01:5070;play=announcement/70000;repeat=1>;tag=0093B4EE-1BA6-1769-9D3A-F900010AAA77-50
Call-ID: 008C88CC-2427-1769-9D01-0F00010AAA77-57@10.1.0.15
CSeq: 1 INVITE
Via: SIP/2.0/UDP 10.1.0.15:5060;branch=z9hG4bK008C89D0-2427-1769-9D01-0F00010AAA77-259;received=10.1.0.15
Contact: <sip:255.255.255.255:5070>
Content-Type: application/sdp
Content-Length: 225

v=0
o=Genesys 50 50 IN IP4 255.255.255.255
s=StreamManager 7.5.005.03 play
c=IN IP4 255.255.255.255
t=0 0
m=audio 8100 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtpmap:0 pcmu/8000

SIPdialog[51] event 14 CALLED/ResOK
generated RTCP(leg=514) at 10:36:51.2589
  RR sender[3bc5d753]
    ssrc[0] frac=0(0.00) lost=0 xseq=0:0 jitter=0 LSR=0 DLSR=2.50
  SDES report
    ssrc[3bc5d753] CNAME='255.255.255.255' TOOL='Genesys SM 7.5.005.03'
generated RTCP(leg=514) at 10:36:56.2594
  RR sender[3bc5d753]
    ssrc[0] frac=0(0.00) lost=0 xseq=0:0 jitter=0 LSR=0 DLSR=7.50
  SDES report
    ssrc[3bc5d753] CNAME='255.255.255.255' TOOL='Genesys SM 7.5.005.03'
generated RTCP(leg=514) at 10:37:01.2613
  RR sender[3bc5d753]
    ssrc[0] frac=0(0.00) lost=0 xseq=0:0 jitter=0 LSR=0 DLSR=12.50
  SDES report
    ssrc[3bc5d753] CNAME='255.255.255.255' TOOL='Genesys SM 7.5.005.03'
generated RTCP(leg=514) at 10:37:06.2584
  RR sender[3bc5d753]
    ssrc[0] frac=0(0.00) lost=0 xseq=0:0 jitter=0 LSR=0 DLSR=17.50
  SDES report
    ssrc[3bc5d753] CNAME='255.255.255.255' TOOL='Genesys SM 7.5.005.03'
generated RTCP(leg=514) at 10:37:11.2614
  RR sender[3bc5d753]
    ssrc[0] frac=0(0.00) lost=0 xseq=0:0 jitter=0 LSR=0 DLSR=22.50
  SDES report
    ssrc[3bc5d753] CNAME='255.255.255.255' TOOL='Genesys SM 7.5.005.03'
generated RTCP(leg=514) at 10:37:16.2639
  RR sender[3bc5d753]
    ssrc[0] frac=0(0.00) lost=0 xseq=0:0 jitter=0 LSR=0 DLSR=27.50
  SDES report
    ssrc[3bc5d753] CNAME='255.255.255.255' TOOL='Genesys SM 7.5.005.03'
Inactivity timeout (30.000 sec) for RTPleg[514]
@10:37:18.7571 [53055] SIPdialog[51] session info 15 (session-timeout)

gsip:CL2CONN[12,UDP]:10:37:18.764 >>>> 394 bytes to 10.1.0.15:5060 >>>>
BYE sip:10.1.0.15:5060 SIP/2.0
From: <sip:annc@LHGPTLDEVGENSMTLX01:5070;play=announcement/70000;repeat=1>;tag=0093B4EE-1BA6-1769-9D3A-F900010AAA77-50
To: "207xxxxxxx" <sip:207xxxxxxx@10.1.0.156>;tag=as1499e728
Call-ID: 008C88CC-2427-1769-9D01-0F00010AAA77-57@10.1.0.15
CSeq: 1 BYE
Content-Length: 0
Via: SIP/2.0/UDP 10.1.0.249:5070;branch=z9hG4bK0093B534-1BA6-1769-9D3A-F900010AAA77-7


SIPdialog[51] event 44 Sent:BYE

gsip:CL2LIST[10,UDP]:10:37:18.777 <<<< 431 bytes from 10.1.0.15:5060 <<<<
SIP/2.0 481 Call Leg/Transaction Does Not Exist
From: <sip:annc@LHGPTLDEVGENSMTLX01:5070;play=announcement/70000;repeat=1>;tag=0093B4EE-1BA6-1769-9D3A-F900010AAA77-50
To: "207xxxxxxx" <sip:207xxxxxxx@10.1.0.156>;tag=as1499e728
Call-ID: 008C88CC-2427-1769-9D01-0F00010AAA77-57@10.1.0.15
CSeq: 1 BYE
Via: SIP/2.0/UDP 10.1.0.249:5070;branch=z9hG4bK0093B534-1BA6-1769-9D3A-F900010AAA77-7;received=10.1.0.249
Content-Length: 0


SIPdialog[51] event 56 TERMINATING/BYE-resp
SIPdialog[51] event 58 DESTROY
  RTPleg[514]:8100/8101 completed (remote 10.1.0.156:19218)
    RX=0/0+0(err=0) rtcp=0(err=0) ssrc[0] jitter=0(max=0)
    TX=0/0+0(err=0) rtcp=6(err=0) seq=0
UDP[99] port 8100 closed at 10:37:18.7822
  file-released(announcement/70000_mulaw.wav)incomplete
  RTPfile[3bc5d753] completed; pkts=0, dropped=0
@10:37:18.7842 [53052] SIPdialog[51] finished (43 annc 0 conf active)


Lumpy

  • Guest
The problem was a stream manager configuration error.  In the Stream Manager application, under options and the "call" section there are two options (call-address and call-protocol).  I had call-address with a value of <ip address:port> when it's just ip address.  So the configuration should be

Section = call
option name 1 = call-address, value = <ip address> e.g. 10.1.0.5
option name 2 = call-protocol, value = sm

Sorry if there were duplicate posts...

Offline cavagnaro

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:) Thanks for posting the detailed solution of your problem, this will help someone else.