Incoming SIP calls dropping after 32 seconds

[TelecomsTim]

We are currently implementing a Server Edition system alongside an Avaya SBC and Hipcom SIP trunks.

After much playing around with the SBC we finally got calls to route in and out however incoming calls are dropping after 32 seconds. We have full speech path during those 32 seconds that the call is connected and outbound calls across the SIP are working perfectly.

As you will see below, the phone system is sending the BYE request and closing the call down.
(I have blanked out the numbers involved in the call for obvious reasons)

18:31:00 97196mS SIP Tx: UDP 192.168.42.1:5060 -> 192.168.42.254:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.42.254:5060;branch=z9hG4bK-s1632-000605106642-1--s1632-
Record-Route: <sip:192.168.42.254:5060;ipcs-line=68;lr;transport=udp>
From: <sip:07XXXXXXXXX@192.168.42.254:5060;user=phone>;tag=SDmcedf01-620307073-1411407087858-
Call-ID: 46e79c65ad4ce119d9c3c81bcef28039
CSeq: 255165562 INVITE
Contact: "44XXXXXXXXX" <sip:44XXXXXXXXX@192.168.42.1:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Server: IP Office 9.0.3.0 build 941
To: "maggies hq" <sip:44XXXXXXXXX@192.168.42.1;user=phone>;tag=18a98279f25e59e4
Content-Type: application/sdp
Content-Length: 203

v=0
o=UserA 962444212 1033067916 IN IP4 192.168.42.1
s=Session SDP
c=IN IP4 192.168.42.1
t=0 0
m=audio 49154 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
18:31:02 99196mS SIP Call Tx: 17
BYE sip:07XXXXXXXXX@192.168.42.254:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.42.1:5060;rport;branch=z9hG4bK8f0751d9ec208fdca144a2915c56fa84
Route: <sip:192.168.42.254:5060;ipcs-line=68;lr;transport=udp>
From: "maggies hq" <sip:44XXXXXXXXX@192.168.42.1;user=phone>;tag=18a98279f25e59e4
To: <sip:07XXXXXXXXX@192.168.42.254;user=phone>;tag=SDmcedf01-620307073-1411407087858-
Call-ID: 46e79c65ad4ce119d9c3c81bcef28039
CSeq: 255165563 BYE
Contact: "44XXXXXXXXX" <sip:44XXXXXXXXX@192.168.42.1:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Reason: Q.850;cause=16;text="Normal call clearing"
User-Agent: IP Office 9.0.3.0 build 941
Content-Length: 0

18:31:02 99196mS SIP Tx: UDP 192.168.42.1:5060 -> 192.168.42.254:5060
BYE sip:07XXXXXXXXX@192.168.42.254:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.42.1:5060;rport;branch=z9hG4bK8f0751d9ec208fdca144a2915c56fa84
Route: <sip:192.168.42.254:5060;ipcs-line=68;lr;transport=udp>
From: "maggies hq" <sip:44XXXXXXXXX@192.168.42.1;user=phone>;tag=18a98279f25e59e4
To: <sip:07XXXXXXXXX@192.168.42.254;user=phone>;tag=SDmcedf01-620307073-1411407087858-
Call-ID: 46e79c65ad4ce119d9c3c81bcef28039
CSeq: 255165563 BYE
Contact: "44XXXXXXXXX" <sip:44XXXXXXXXX@192.168.42.1:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer
Reason: Q.850;cause=16;text="Normal call clearing"
User-Agent: IP Office 9.0.3.0 build 941
Content-Length: 0

18:31:02 99196mS Sip: SIPDialog f0627958 deleted, dialogs 0
18:31:02 99196mS CMCallEvt: 17.1004.1 1 SIPTrunk Endpoint: StateChange: END=A CMCSOGConnReq->CMCSCompleted
18:31:02 99196mS CMExtnTx: v=3501, p1=0



I also came across this on a Askerisk forum which pretty much matches my exact issue but I wondered in any of you guys have seen this on an IPO and how you resolved it.

http://forums.asterisk.org/viewtopic.php?f=1&t=86149

I have an IP route setup on the system to route the call to the SIP provider using the SBC as the gateway.

We also have an issue where we receive silence when calling into voicemail which is likley to be linked to the issue above.


Any support would be kindly appreciated

 

[FFrey]

Are you using direct media path between the IPO and SBC?

 

[TelecomsTim]

I've tried both and made no difference

 

[FFrey]

Are you using LAN2 as a dedicated interface to the SBC? If your using re-invite turn it off as I have seen that cause issues in the past. I can only assume your using the same compression on both ends right? You could try turning on keepalives.

 

[TelecomsTim]

I've been doing some reading over night and it seems this is quite common in the world of SIP trunking. The issue seems to be the Avaya not receiving the ACK message from the SIP provider and therefore dropping the call. Others have suggested the RTP ports are being blocked so I will check that this morning.

Still not solved the issue but at least I think I know what is causing the problem. Please continue to come forward with any suggestions.

Thanks

 

[jamie77]

Very common and always the router not doing its job. It could be the SBC in your case but I have no knowledge of them to say either way.

We had a small issue with ours today after a router reboot (31 sec drop on i/c, no o/g). Disabling and re-enabling SIP ALG on the Sonicwall fixed it straight away.

the 32 sec timer is the network not getting responses so assume the IPO doesn't have the call, so it drops it.

Are you using ALG or STUN on the IPO/SBC??

Jamie Green

[bold]A[/bold]vaya [bold]R[/bold]egistered [bold]S[/bold]pecialist [bold]E[/bold]ngineer