SPA2102 Call Drops

Unanswered Question
May 26th, 2010

Here is our problem with the SPA2102.  Please take a look at it.  If you have a workable solution let me know.

When I make an outbound call from the Linksys SPA2102 to any number the any number has the call dropped after the call has been connected for roughly 23 seconds.  From the phone Linksys SPA2102 the call thinks it is still going and doesn't seem to end.

I have included my conversation regarding the issue with our Metaswitch CSE.  I have also included a wireshark capture of the call from between the SBG901 and the SPA2102 and a call trace of a dropped call from the Metaswitch.  Firmware used in the test is 5.2.10, though the SPA2102s in the field are all 5.1.12 which is what is certified with our version of the Metaswitch.  The Motorola modem in this case is running the most current version of firmware available from Motorola.

From our Metaswitch is sent a re-invite, which is done every 30 seconds from the initial invite (to verify the call is up and still connected), and the device sends us a 481 Does not exist, so the switch kills the call.

Here is the trace from our switch:

17:18:56.733 49341584 20122047:0001   SIP >> INVITE  - DN: 6084934158 Call: 0 Trx: ce4f70
17:18:56.733 49341584 20122047:0001   SIP    Content-Length: 164
17:18:56.733 49341584 20122047:0001   SIP    Content-Type: application/sdp
17:18:56.733 49341584 20122047:0001   SIP    v=0
17:18:56.733 49341584 20122047:0001   SIP    o=- 189354392 189354392 IN IP4 209.83.10.25
17:18:56.733 49341584 20122047:0001   SIP    s=-
17:18:56.733 49341584 20122047:0001   SIP    c=IN IP4 209.83.10.25
17:18:56.733 49341584 20122047:0001   SIP    t=0 0
17:18:56.733 49341584 20122047:0001   SIP    m=audio 33884 RTP/AVP 18 0 101
17:18:56.733 49341584 20122047:0001   SIP    a=rtpmap:101 telephone-event/8000
17:18:56.733 49341584 20122047:0001   SIP    a=ptime:20
17:18:56.763 49341584 20122047:0001   SIP << 481 Call Leg/Transaction Does Not Exist - DN: 6084934158 Call: 0 Trx: ce4f70
17:18:56.763 49341584 20122047:0001   SIP    To: 6084934158<sip:[email protected]>;tag=fb92de4d
17:18:56.763 49341584 20122047:0001   SIP    From: <sip:[email protected]>;tag=ms.merr.com+1+448163+b915a9
17:18:56.763 49341584 20122047:0001   SIP    Via: SIP/2.0/UDP ms.merr.com;branch=z9hG4bK-b18e498378cfe8cf06cade1d17f2f5e7-ms
17:18:56.763 49341584 20122047:0001   SIP      .merr.com-1
17:18:56.763 49341584 20122047:0001   SIP    Call-ID: 0c0af54a005c49a4622ae6832b66e52d
17:18:56.763 49341584 20122047:0001   SIP    CSeq: 215713198 INVITE
17:18:56.763 49341584 20122047:0001   SIP    Contact: 6084934158<sip:[email protected]
17:18:56.763 49341584 20122047:0001   SIP      83.10.45:5060>
17:18:56.763 49341584 20122047:0001   SIP    Server: Linksys/SPA2102-5.2.10 (000e081c07ef)
17:18:56.763 49341584 20122047:0001   SIP    Content-Length: 0 
I have this problem too.
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Alberto Montilla Wed, 05/26/2010 - 09:30

Dear Sir;

Can you send the configuration of the SPA2102? I'm interested to see the RTP Packet Size parameter on tab (provide this value instead of the whole config file please). If it is set to 0.030, please change it to 0.020 and apply changes (unit should reboot). Try again the call. May be a mismatch of packetization time.

Regards

Alberto

benjamincone Wed, 05/26/2010 - 15:21


GMT-06:00
-3
-3
no
no
Yes
Yes
68.216.79.113
64.236.96.53
very  high
disable
no
no
no
no
no
no
no
no
no
Sinusoid
25
16
70
440@-10
1
Yes
Yes
Yes
Yes
Yes
2
Parallel
pool.ntp.org
Yes
Yes
10
3600
14400
Yes
Yes
5
2
$VERSION  ($MA)
$VERSION ($MA)
application/dtmf-relay
application/hook-flash
No
No
.5
4
5
32
32
32
32
32
240
30
15
7200
16384
16482
0.030
0
0
100
101
98
97
96
99
NSE
telephone-event
PCMU
PCMA
G726-16
G726-24
G726-32
G726-40
G729a
G729ab
G723
No
No
No
No
No
No
No
15
No
30
Yes
$NOTIFY
$PROXY
0x68
0xb8
No
Yes
Yes
none
Yes
Yes
Yes
Yes
No
Yes
600
no
no
no
no
No
No
No
No
Yes
No
No
Yes
Yes
Yes
Yes
Yes
No
Yes
Yes
Yes
Yes
Yes
G711u
Yes
ReINVITE
AVT
Yes
Yes
No
Forward
Forward
Forward
No
1
0
New VM  Arrives
No
Yes
No
30
yes
$NOTIFY
$PROXY
0x68
0xb8
no
no
no
no
no
no
no
no
no
no
ReINVITE
Yes
3
#98
#99
#90
#91
#92
#93

That is pretty much what we have configured, plus of course line information, codec, and a couple of other things.  I did try manually changing the RTP Packet Size to what you recommended, but it was the same problem.  The ATA itself doesn't realize that the call dropped and just hangs in silence.

Alberto Montilla Wed, 05/26/2010 - 16:19

Dear Sir;

Please take traces on the ATA . I need to look at the trace, it may have to do with ATA not receiving the RTP stream or a SIP message (potentially because of an ALG issue on the router).


Regards
Alberto

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