SIP client dropping after 20 seconds with DisconnectCause 66 Error

johns · ‎06-21-2012

When a specific SIP client connects to the 1760 Gateway we recieve a disconnectText "Recovery on timer expiry (102). I am using CAS signaling and not ISDN. How do I set timer t310 or what is the equivalent for CAS? I am on software version 12.2 and don't have ISDN enabled.

Other SIP clients have no problem and terminate correctly.

I got the error message from the syscon log generated by the gateway.

Daniele Giordano · ‎06-22-2012

Can you add the output of "debug ccsip messages"?

Regards.

johns · ‎06-22-2012

c=IN IP4 10.1.10.119

*Feb 10 23:51:06.134: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:%2301050254@10.1.10.119:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.120.117:5071;branch=z9hG4bK-d8754z-68675d1e130e7c38-1---d8754z-;rport
Max-Forwards: 70
Contact:
To: <>;tag=F44A4E9-2297
From: "Jeff";tag=0d2efe10
Call-ID: M2IxYWNmNGQ5NGE5M2FmMDBiNmQxMWEyYjFkYzU3M2Q.
CSeq: 1 ACK
User-Agent: ABTO Video SIP SDK
Content-Length: 0

*Feb 10 23:51:08.129: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.120.117:5071;branch=z9hG4bK-d8754z-a454081b993d4927-1---d8754z-;rport
From: "Jeff";tag=0d2efe10
To: <>;tag=F44A4E9-2297
Date: Fri, 10 Feb 2012 23:51:02 GMT
Call-ID: M2IxYWNmNGQ5NGE5M2FmMDBiNmQxMWEyYjFkYzU3M2Q.
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1 INVITE
Session-Expires: 1800;refresher=uac
Require: timer
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <>
Content-Type: application/sdp
Content-Length: 282

v=0
o=CiscoSystemsSIP-GW-UserAgent 5622 8647 IN IP4 10.1.10.119
s=SIP Call
c=IN IP4 10.1.10.119
t=0 0
m=audio 17218 RTP/AVP 0 99
c=IN IP4 10.1.10.119
a=rtpmap:0 PCMU/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=ptime:20
m=video 0 RTP/AVP
c=IN IP4 10.1.10.119

*Feb 10 23:51:08.137: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:%2301050254@10.1.10.119:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.120.117:5071;branch=z9hG4bK-d8754z-68675d1e130e7c38-1---d8754z-;rport
Max-Forwards: 70
Contact:
To: <>;tag=F44A4E9-2297
From: "Jeff";tag=0d2efe10
Call-ID: M2IxYWNmNGQ5NGE5M2FmMDBiNmQxMWEyYjFkYzU3M2Q.
CSeq: 1 ACK
User-Agent: ABTO Video SIP SDK
Content-Length: 0

*Feb 10 23:51:12.132: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.120.117:5071;branch=z9hG4bK-d8754z-a454081b993d4927-1---d8754z-;rport
From: "Jeff";tag=0d2efe10
To: <>;tag=F44A4E9-2297
Date: Fri, 10 Feb 2012 23:51:02 GMT
Call-ID: M2IxYWNmNGQ5NGE5M2FmMDBiNmQxMWEyYjFkYzU3M2Q.
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1 INVITE
Session-Expires: 1800;refresher=uac
Require: timer
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <>
Content-Type: application/sdp
Content-Length: 282

v=0
o=CiscoSystemsSIP-GW-UserAgent 5622 8647 IN IP4 10.1.10.119
s=SIP Call
c=IN IP4 10.1.10.119
t=0 0
m=audio 17218 RTP/AVP 0 99
c=IN IP4 10.1.10.119
a=rtpmap:0 PCMU/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=ptime:20
m=video 0 RTP/AVP
c=IN IP4 10.1.10.119

*Feb 10 23:51:12.140: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:%2301050254@10.1.10.119:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.120.117:5071;branch=z9hG4bK-d8754z-68675d1e130e7c38-1---d8754z-;rport
Max-Forwards: 70
Contact:
To: <>;tag=F44A4E9-2297
From: "Jeff";tag=0d2efe10
Call-ID: M2IxYWNmNGQ5NGE5M2FmMDBiNmQxMWEyYjFkYzU3M2Q.
CSeq: 1 ACK
User-Agent: ABTO Video SIP SDK
Content-Length: 0

*Feb 10 23:51:16.135: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.120.117:5071;branch=z9hG4bK-d8754z-a454081b993d4927-1---d8754z-;rport
From: "Jeff";tag=0d2efe10
To: <>;tag=F44A4E9-2297
Date: Fri, 10 Feb 2012 23:51:02 GMT
Call-ID: M2IxYWNmNGQ5NGE5M2FmMDBiNmQxMWEyYjFkYzU3M2Q.
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1 INVITE
Session-Expires: 1800;refresher=uac
Require: timer
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <>
Content-Type: application/sdp
Content-Length: 282

v=0
o=CiscoSystemsSIP-GW-UserAgent 5622 8647 IN IP4 10.1.10.119
s=SIP Call
c=IN IP4 10.1.10.119
t=0 0
m=audio 17218 RTP/AVP 0 99
c=IN IP4 10.1.10.119
a=rtpmap:0 PCMU/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=ptime:20
m=video 0 RTP/AVP
c=IN IP4 10.1.10.119

*Feb 10 23:51:16.179: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:%2301050254@10.1.10.119:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.120.117:5071;branch=z9hG4bK-d8754z-68675d1e130e7c38-1---d8754z-;rport
Max-Forwards: 70
Contact:
To: <>;tag=F44A4E9-2297
From: "Jeff";tag=0d2efe10
Call-ID: M2IxYWNmNGQ5NGE5M2FmMDBiNmQxMWEyYjFkYzU3M2Q.
CSeq: 1 ACK
User-Agent: ABTO Video SIP SDK
Content-Length: 0

*Feb 10 23:51:20.137: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.120.117:5071;branch=z9hG4bK-d8754z-a454081b993d4927-1---d8754z-;rport
From: "Jeff";tag=0d2efe10
To: <>;tag=F44A4E9-2297
Date: Fri, 10 Feb 2012 23:51:02 GMT
Call-ID: M2IxYWNmNGQ5NGE5M2FmMDBiNmQxMWEyYjFkYzU3M2Q.
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1 INVITE
Session-Expires: 1800;refresher=uac
Require: timer
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <>
Content-Type: application/sdp
Content-Length: 282

v=0
o=CiscoSystemsSIP-GW-UserAgent 5622 8647 IN IP4 10.1.10.119
s=SIP Call
c=IN IP4 10.1.10.119
t=0 0
m=audio 17218 RTP/AVP 0 99
c=IN IP4 10.1.10.119
a=rtpmap:0 PCMU/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=ptime:20
m=video 0 RTP/AVP
c=IN IP4 10.1.10.119

*Feb 10 23:51:20.161: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:%2301050254@10.1.10.119:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.120.117:5071;branch=z9hG4bK-d8754z-68675d1e130e7c38-1---d8754z-;rport
Max-Forwards: 70
Contact:
To: <>;tag=F44A4E9-2297
From: "Jeff";tag=0d2efe10
Call-ID: M2IxYWNmNGQ5NGE5M2FmMDBiNmQxMWEyYjFkYzU3M2Q.
CSeq: 1 ACK
User-Agent: ABTO Video SIP SDK
Content-Length: 0

*Feb 10 23:51:24.140: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 2, ConnectionId EC3F0FE3 537811E1 83AB9DCE F0C03199, SetupTime *15:51:02.270 pdt
Fri Feb 10 2012, PeerAddress Jeff, PeerSubAddress , DisconnectCause 66 , DisconnectText recovery on timer expiry (102), ConnectTime *15:51:04.62
0 pdt Fri Feb 10 2012, DisconnectTime *15:51:24.140 pdt Fri Feb 10 2012, CallOrigin 2, ChargedUnits 0, InfoType 2, TransmitPackets 976, TransmitB
ytes 156160, ReceivePackets 969, ReceiveBytes 154888
*Feb 10 23:51:24.969: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1, ConnectionId EC3F0FE3 537811E1 83AB9DCE F0C03199, SetupTime *15:51:02.279 pdt
Fri Feb 10 2012, PeerAddress #01050254, PeerSubAddress , DisconnectCause 66 , DisconnectText recovery on timer expiry (102), ConnectTime *15:51:
04.599 pdt Fri Feb 10 2012, DisconnectTime *15:51:24.969 pdt Fri Feb 10 2012, CallOrigin 1, ChargedUnits 0, InfoType 2, TransmitPackets 969, Tran
smitBytes 154888, ReceivePackets 976, ReceiveBytes 156160
*Feb 10 23:51:41.360: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:%2301050254@10.1.10.119:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.120.117:5071;branch=z9hG4bK-d8754z-b31b02582d7c7362-1---d8754z-;rport
Max-Forwards: 70
Contact:
To: <>;tag=F44A4E9-2297
From: "Jeff";tag=0d2efe10
Call-ID: M2IxYWNmNGQ5NGE5M2FmMDBiNmQxMWEyYjFkYzU3M2Q.
CSeq: 3 BYE
User-Agent: ABTO Video SIP SDK
Content-Length: 0

*Feb 10 23:51:41.364: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.0.120.117:5071;branch=z9hG4bK-d8754z-b31b02582d7c7362-1---d8754z-;rport
From: "Jeff";tag=0d2efe10
To: <>;tag=F44A4E9-2297
Call-ID: M2IxYWNmNGQ5NGE5M2FmMDBiNmQxMWEyYjFkYzU3M2Q.
CSeq: 3 BYE
Content-Length: 0

*Feb 10 23:51:41.636: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
REGISTER sip:10.1.10.119 SIP/2.0
Via: SIP/2.0/UDP 10.0.120.164:5060;rport;branch=z9hG4bK88139128
Max-Forwards: 70
To:
From: ;tag=9727
Call-ID: 1339712787-9128-JOHNNYTC@10.0.120.164
CSeq: 3 REGISTER
Contact: ;expires=3600;q=0.90
User-Agent: NCH Software Express Talk 4.26
Content-Length: 0

*Feb 10 23:51:41.644: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.0.120.164:5060;rport;branch=z9hG4bK88139128
From: ;tag=9727
To:
Date: Fri, 10 Feb 2012 23:51:41 GMT
Call-ID: 1339712787-9128-JOHNNYTC@10.0.120.164
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 3 REGISTER

*Feb 10 23:51:56.380: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
REGISTER sip:10.1.10.119 SIP/2.0
Via: SIP/2.0/UDP 10.0.120.117:5071;branch=z9hG4bK-d8754z-eb3b5a3e8a20b70a-1---d8754z-;rport
Max-Forwards: 70
Contact: ;methods="INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE";expi
res=0
To: "Jeff"
From: "Jeff";tag=2d215c73
Call-ID: NTQ4YmRjYWM3MjkxZmVkM2RjZDg2NmUzNDA2M2VmMjU.
CSeq: 2 REGISTER
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, SUBSCRIBE, INFO, MESSAGE
Supported: replaces, timer, norefersub, answermode, tdialog
User-Agent: ABTO Video SIP SDK
Content-Length: 0

*Feb 10 23:51:56.388: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.0.120.117:5071;branch=z9hG4bK-d8754z-eb3b5a3e8a20b70a-1---d8754z-;rport
From: "Jeff";tag=2d215c73
To: "Jeff"
Date: Fri, 10 Feb 2012 23:51:56 GMT
Call-ID: NTQ4YmRjYWM3MjkxZmVkM2RjZDg2NmUzNDA2M2VmMjU.
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 2 REGISTER

Daniele Giordano · ‎06-23-2012

In your debug is missing the initial phase of setup: INVITE from ABTO Video SIP SDK.

There are only a 200 OK sent from your Cisco and ACK sent from ABTO Video SIP SDK and received from the Cisco.

In 200 OK messages there are audio and video capabilities and the option "Require: timer".

Can you check the initial INVITE? What capabilities are present? Is there the timer options?

The 200 OK - ACK phase should be correct and sufficient to start the audio/video stream. Seems that the Cisco doesn't accept the ACK.

Can you upgrade the IOS with a newer one (12.3 or 12.4)?

Can you add a full "debug ccsip all" ?

Regards.

johns · ‎06-25-2012

I beleive this is the initial invite.

VOIPGateway#termino al monitor
VOIPGateway#
Jun 25 22:48:43.291: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:%2301050220@10.1.10.119 SIP/2.0

Via: SIP/2.0/UDP 10.0.120.117:5071;branch=z9hG4bK-d8754z-7938c672427d7e6d-1---d8754z-;rport

Max-Forwards: 70

Contact:

To: <>

From: "Jeff";tag=540e5275

Call-ID: YWE5OTRjNjY0NjU4YzRiMWQxNTBhM2E3ZWUwMzg3M2U.

CSeq: 1 INVITE

Session-Expires: 1800

Min-SE: 90

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, SUBSCRIBE, INFO, MESSAGE

Content-Type: application/sdp

Supported: replaces, timer, norefersub, answermode, tdialog, 100rel

User-Agent: ABTO Video SIP SDK

Content-Length: 456

v=0

o=- 12985134615020019 12985134615020019 IN IP4 10.0.120.117

s=-

c=IN IP4 10.0.120.117

t=0 0

m=audio 17440 RTP/AVP 0 8 96 97 98 99

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:96 speex/8000

a=rtpmap:97 ilbc/8000

a=rtpmap:98 ilbc/8000

a=fmtp:98 mode=20

a=rtpmap:99 telephone-event/8000

a=fmtp:99 0-15

a=ptime:20

a=sendrecv

m=video 17442 RTP/AVP 120

a=rtpmap:120 h263-1998/90000

a=fmtp:120 CIF4=1;CIF=1;QCIF=1;SQCIF=1

a=sendrecv

Jun 25 22:48:43.323: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 10.0.120.117:5071;branch=z9hG4bK-d8754z-7938c672427d7e6d-1---d8754z-;rport

From: "Jeff";tag=540e5275

To: <>;tag=1FB006DC-1CF3

Date: Mon, 25 Jun 2012 22:48:43 GMT

Call-ID: YWE5OTRjNjY0NjU4YzRiMWQxNTBhM2E3ZWUwMzg3M2U.

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 1 INVITE

Allow-Events: telephone-event

Content-Length: 0

Jun 25 22:48:45.651: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 10.0.120.117:5071;branch=z9hG4bK-d8754z-7938c672427d7e6d-1---d8754z-;rport

From: "Jeff";tag=540e5275

To: <>;tag=1FB006DC-1CF3

Date: Mon, 25 Jun 2012 22:48:43 GMT

Call-ID: YWE5OTRjNjY0NjU4YzRiMWQxNTBhM2E3ZWUwMzg3M2U.

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 1 INVITE

Require: 100rel

RSeq: 207

Allow: UPDATE

Allow-Events: telephone-event

Contact: <>

Content-Disposition: session;handling=required

Content-Type: application/sdp

Content-Length: 282

v=0

o=CiscoSystemsSIP-GW-UserAgent 6564 3608 IN IP4 10.1.10.119

s=SIP Call

c=IN IP4 10.1.10.119

t=0 0

m=audio 17100 RTP/AVP 0 99

c=IN IP4 10.1.10.119

a=rtpmap:0 PCMU/8000

a=rtpmap:99 telephone-event/8000

a=fmtp:99 0-15

a=ptime:20

m=video 0 RTP/AVP

c=IN IP4 10.1.10.119

Daniele Giordano · ‎06-26-2012

Video negotiation is not correct:

INVITE from ABTO Video SIP SDK offers

m=audio 17440 RTP/AVP 0 8 96 97 98 99

0 pcmu/8000

8 pcma/8000

96 speex/8000

97 ilbc/8000

98 ilbc/8000

99 telephone-event/8000

m=video 17442 RTP/AVP 120

120 h263

but your cisco responds for audio codec only

m=audio 17100 RTP/AVP 0 99

0 PCMU/8000

99 telephone-event/8000

m=video 0 RTP/AVP

Your cisco requires "100rel" and "timer" support.

Try to disable video capability from your ABTO Video SIP SDK and disable 100rel adn timer from cisco.

Regards.