[cisco-voip] 78xx phones in SRST mode - Stops working after about 20-30 minutes, other phone models continue to work
Brian V
bvanbens at gmail.com
Wed Dec 21 12:57:11 EST 2016
I've having a strange issue with 78xx phones while in SRST mode.
I've isolated it on the bench and I'm able to reproduce it.
I have a TAC case open but wondering if anyone has run into anything
similar?
have 3 phones for testing.
7821 SIP phone load: 11-0-1-11
7965 SCCP phone (SCCP45.9-4-2SR1)
8961 SIP phone - older one but with newer code (sip8961.9.4.2 SR2)
Router/IOS: 4321 - Denali 16.03.02
CUCM version 10.5.2 SU2
Typical remote CUCM site configured for SRST. Mix of 7821, 8961, and 7965
phones.
Router is configured for both SCCP and SIP SRST.
When WAN connection is lost, phones register with router correctly after a
few minutes and function properly. Can make calls between them all, and
calls to/from PSTN
After a period of time (about 20-30 min) the 78xx phones seem to stop
working. Can't make calls TO them and can't make calls FROM them. During
this time calls work between the 8961 phones and the 7965 phones and
to/from the PSTN.
During this time where the 7821 phones don’t work, the output of the *show
voice register* commands shows the 7821 phone is registered with the
auto-created dial-peers still present.
I enabled SIP message debugging, this is where it gets interesting.
Test is to place a call TO the 7821 phone. From SCCP ext. 2014 to SIP(7821
phones) ext. 2004
I placed a call from the SCCP phone to eliminate one side of the SIP
messages but the result is the same regardless if it place the call from an
8961 or 79xx phone.
I see the immediately router send out 3 invites to the 7821 phone
During this time, the phone doesn’t show any activity.
Then about 1 full minute later we see the 7821 phone send back three
100-Trying messages and three 180-ringing messages
During this time the 7821 phone is now ringing a full minute after I
already hung up the calling phone
I pick up the handset on the 7821 phone, there is not an actual call there.
I hang it up
Then about 2 minutes later the 7821 phone sends back 200-OK messages with
SDP. There was no activity on the phone during this time.
It almost "seems" like the 7821 phone is having high CPU usage and its
super slow in responding to SIP messages.
This state take a while to develop. Immediately after the phones switch
into SRST mode they work fine , then after a period of time, just the 78xx
phones exhibit this behavior.
See debugs and config below
SRST IP: 172.25.4.200
Target phone IP: 172.25.4.26
voice service voip
ip address trusted list
ipv4 10.0.0.0 255.0.0.0
ipv4 192.168.0.0 255.255.0.0
ipv4 10.10.111.10 255.255.255.255
ipv4 172.16.0.0 255.240.0.0
media statistics
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
call-quality
sip
bind control source-interface GigabitEthernet0/0/0
bind media source-interface GigabitEthernet0/0/0
min-se 600
registrar server expires max 1800 min 900
no call service stop
voice register global
default mode
no allow-hash-in-dn
timeouts interdigit 5
system message SIP SRST Mode
max-dn 25
max-pool 10
timezone 8
!
voice register pool 1
id network 172.25.4.0 mask 255.255.255.0
dtmf-relay rtp-nte sip-kpml
codec g711ulaw
no vad
!
voice hunt-group 1 parallel
list 2004,3003,2001,2014
pilot 2424
interface GigabitEthernet0/0/0
ip address 172.25.4.200 255.255.255.0
media-type rj45
negotiation auto
call-manager-fallback
secondary-dialtone 9
max-conferences 8 gain -6
transfer-system full-consult
timeouts interdigit 5
ip source-address 172.25.4.200 port 2000
max-ephones 10
max-dn 25 octo-line
system message primary SCCP SRST MODE
time-zone 8
TEST-CUBE#
TEST-CUBE# sh debug
IOSXE Conditional Debug Configs:
Conditional Debug Global State: Stop
IOSXE Packet Tracing Configs:
Packet Infra debugs:
Ip Address Port
------------------------------------------------------|----------
CCSIP SPI: SIP Call Message tracing is enabled (filter is OFF)
CCSIP SPI: SIP Out-of-Dialog tracing is enabled (filter is OFF)
voice-register:
voice-register error debugging is on
voice-register events debugging is on
TEST-CUBE#
!! ## this event is about 20-30 min after the phone went into SRST mode and
registered with the router
!! ## prior to this time, it functioned normally.
!!#### make call from SCCP phone extension 2001 to SIP phone extension 2004
!!### Sends 3 invites is rapid succession. Note time stamps
030590: .Dec 20 *17:38:16.801*: //3480/365A43AF9107/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:2004 at 172.25.4.26:5060 SIP/2.0
Via: SIP/2.0/UDP 172.25.4.200:5060;branch=z9hG4bKB871399
Remote-Party-ID: "+12025552001" <sip:2001 at 172.25.4.200
>;party=calling;screen=no;privacy=off
From: "+12025552001" <sip:2001 at 172.25.4.200>;tag=5E6EC01-2547
To: <sip:2004 at 172.25.4.26>
Date: Tue, 20 Dec 2016 23:38:16 GMT
Call-ID: 3739F3E8-C64411E6-910CD720-9D491D8F at 172.25.4.200
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 600
Cisco-Guid: 0911885231-3326349798-2433210144-2638814607
User-Agent: Cisco-SIPGateway/IOS-16.3.2
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1482277096
Contact: <sip:2001 at 172.25.4.200:5060>
Expires: 180
Allow-Events: kpml, telephone-event
Session-ID:
e3d1d5b9cf5d5821bc81fbad4de46749;remote=00000000000000000000000000000000
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 246
v=0
o=CiscoSystemsSIP-GW-UserAgent 4428 5277 IN IP4 172.25.4.200
s=SIP Call
c=IN IP4 172.25.4.200
t=0 0
m=audio 8508 RTP/AVP 0 101
c=IN IP4 172.25.4.200
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
030591: .Dec 20 17:38:17.051: //3480/365A43AF9107/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:2004 at 172.25.4.26:5060 SIP/2.0
Via: SIP/2.0/UDP 172.25.4.200:5060;branch=z9hG4bKB871399
Remote-Party-ID: "+12025552001" <sip:2001 at 172.25.4.200
>;party=calling;screen=no;privacy=off
From: "+12025552001" <sip:2001 at 172.25.4.200>;tag=5E6EC01-2547
To: <sip:2004 at 172.25.4.26>
Date: Tue, 20 Dec 2016 23:38:17 GMT
Call-ID: 3739F3E8-C64411E6-910CD720-9D491D8F at 172.25.4.200
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 600
Cisco-Guid: 0911885231-3326349798-2433210144-2638814607
User-Agent: Cisco-SIPGateway/IOS-16.3.2
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1482277097
Contact: <sip:2001 at 172.25.4.200:5060>
Expires: 180
Allow-Events: kpml, telephone-event
Session-ID:
e3d1d5b9cf5d5821bc81fbad4de46749;remote=00000000000000000000000000000000
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 246
v=0
o=CiscoSystemsSIP-GW-UserAgent 4428 5277 IN IP4 172.25.4.200
s=SIP Call
c=IN IP4 172.25.4.200
t=0 0
m=audio 8508 RTP/AVP 0 101
c=IN IP4 172.25.4.200
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
030592: .Dec 20 17:38:17.550: //3480/365A43AF9107/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:2004 at 172.25.4.26:5060 SIP/2.0
Via: SIP/2.0/UDP 172.25.4.200:5060;branch=z9hG4bKB871399
Remote-Party-ID: "+12025552001" <sip:2001 at 172.25.4.200
>;party=calling;screen=no;privacy=off
From: "+12025552001" <sip:2001 at 172.25.4.200>;tag=5E6EC01-2547
To: <sip:2004 at 172.25.4.26>
Date: Tue, 20 Dec 2016 23:38:17 GMT
Call-ID: 3739F3E8-C64411E6-910CD720-9D491D8F at 172.25.4.200
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 600
Cisco-Guid: 0911885231-3326349798-2433210144-2638814607
User-Agent: Cisco-SIPGateway/IOS-16.3.2
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1482277097
Contact: <sip:2001 at 172.25.4.200:5060>
Expires: 180
Allow-Events: kpml, telephone-event
Session-ID:
e3d1d5b9cf5d5821bc81fbad4de46749;remote=00000000000000000000000000000000
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 246
v=0
o=CiscoSystemsSIP-GW-UserAgent 4428 5277 IN IP4 172.25.4.200
s=SIP Call
c=IN IP4 172.25.4.200
t=0 0
m=audio 8508 RTP/AVP 0 101
c=IN IP4 172.25.4.200
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
030593: .Dec 20 17:38:18.550: %VOICE_IEC-3-GW: SIP: Internal Error (1xx
wait timeout): IEC=1.1.129.7.59.0 on callID 3480
GUID=365A43AFC64411E69107D7209D491D8F
030594: .Dec 20 17:38:18.552: voice_reg_get_phone_index_from_contact:
Trying to match key 2004
030595: .Dec 20 17:38:18.552: voice_reg_get_phone_index_from_contact:key
MATCHED!!
!!!### We receive the first of three trying responses at 17:39:31, almost
1 full minute from the last invite message at 17:38:17
!!! the called phone rings at this point with a "phantom" call. and I
picked up the handset
030596: .Dec 20 17:39:31.723: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.25.4.200:5060;branch=z9hG4bKB871399
From: "+12025552001" <sip:2001 at 172.25.4.200>;tag=5E6EC01-2547
To: <sip:2004 at 172.25.4.26>
Call-ID: 3739F3E8-C64411E6-910CD720-9D491D8F at 172.25.4.200
Session-ID:
0ca99dec00105000a0006899cd85aa57;remote=e3d1d5b9cf5d5821bc81fbad4de46749
Date: Tue, 20 Dec 2016 23:39:30 GMT
CSeq: 101 INVITE
Server: Cisco-CP7821/11.0.1
Contact: <sip:445806ff-0c30-04cf-f8ad-3fb72b5119c9 at 172.25.4.26:5060
;transport=udp>;+u.sip!devicename.ccm.cisco.com="SEP6899CD85AA57"
Allow:
ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Supported:
replaces,join,sdp-anat,norefersub,resource-priority,X-cisco-srtp-fallback,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 0
030597: .Dec 20 17:39:31.761: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.25.4.200:5060;branch=z9hG4bKB871399
From: "+12025552001" <sip:2001 at 172.25.4.200>;tag=5E6EC01-2547
To: <sip:2004 at 172.25.4.26>
Call-ID: 3739F3E8-C64411E6-910CD720-9D491D8F at 172.25.4.200
Session-ID:
0ca99dec00105000a0006899cd85aa57;remote=e3d1d5b9cf5d5821bc81fbad4de46749
Date: Tue, 20 Dec 2016 23:39:30 GMT
CSeq: 101 INVITE
Server: Cisco-CP7821/11.0.1
Contact: <sip:445806ff-0c30-04cf-f8ad-3fb72b5119c9 at 172.25.4.26:5060
;transport=udp>;+u.sip!devicename.ccm.cisco.com="SEP6899CD85AA57"
Allow:
ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Supported:
replaces,join,sdp-anat,norefersub,resource-priority,X-cisco-srtp-fallback,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 0
030599: .Dec 20 17:39:31.764: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.25.4.200:5060;branch=z9hG4bKB871399
From: "+12025552001" <sip:2001 at 172.25.4.200>;tag=5E6EC01-2547
To: <sip:2004 at 172.25.4.26>
Call-ID: 3739F3E8-C64411E6-910CD720-9D491D8F at 172.25.4.200
Session-ID:
0ca99dec00105000a0006899cd85aa57;remote=e3d1d5b9cf5d5821bc81fbad4de46749
Date: Tue, 20 Dec 2016 23:39:30 GMT
CSeq: 101 INVITE
Server: Cisco-CP7821/11.0.1
Contact: <sip:445806ff-0c30-04cf-f8ad-3fb72b5119c9 at 172.25.4.26:5060
;transport=udp>;+u.sip!devicename.ccm.cisco.com="SEP6899CD85AA57"
Allow:
ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Supported:
replaces,join,sdp-anat,norefersub,resource-priority,X-cisco-srtp-fallback,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 0
030600: .Dec 20 17:39:31.818: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.25.4.200:5060;branch=z9hG4bKB871399
From: "+12025552001" <sip:2001 at 172.25.4.200>;tag=5E6EC01-2547
To: <sip:2004 at 172.25.4.26>;tag=6899cd85aa57010c52826a47-3bf1086e
Call-ID: 3739F3E8-C64411E6-910CD720-9D491D8F at 172.25.4.200
Session-ID:
0ca99dec00105000a0006899cd85aa57;remote=e3d1d5b9cf5d5821bc81fbad4de46749
Date: Tue, 20 Dec 2016 23:39:30 GMT
CSeq: 101 INVITE
Server: Cisco-CP7821/11.0.1
Contact: <sip:445806ff-0c30-04cf-f8ad-3fb72b5119c9 at 172.25.4.26:5060
;transport=udp>;+u.sip!devicename.ccm.cisco.com="SEP6899CD85AA57"
Allow:
ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "Candace Flynn" <sip:2004 at 172.25.4.200
>;party=called;id-type=subscriber;privacy=off;screen=yes
Supported:
replaces,join,sdp-anat,norefersub,resource-priority,X-cisco-srtp-fallback,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 0
!!!! ### phones sends back OK message at 17:41:31, almost 2 minutes after
it sent back the 180 Ringing and I picked up handset
030603: .Dec 20 17:41:31.828: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.25.4.200:5060;branch=z9hG4bKB871399
From: "+12025552001" <sip:2001 at 172.25.4.200>;tag=5E6EC01-2547
To: <sip:2004 at 172.25.4.26>;tag=6899cd85aa57010c52826a47-3bf1086e
Call-ID: 3739F3E8-C64411E6-910CD720-9D491D8F at 172.25.4.200
Session-ID:
0ca99dec00105000a0006899cd85aa57;remote=e3d1d5b9cf5d5821bc81fbad4de46749
Date: Tue, 20 Dec 2016 23:41:30 GMT
CSeq: 101 INVITE
Server: Cisco-CP7821/11.0.1
Contact: <sip:445806ff-0c30-04cf-f8ad-3fb72b5119c9 at 172.25.4.26:5060
;transport=udp>;+u.sip!devicename.ccm.cisco.com="SEP6899CD85AA57"
Allow:
ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "Candace Flynn" <sip:2004 at 172.25.4.200
>;party=called;id-type=subscriber;privacy=off;screen=yes
Supported:
replaces,join,sdp-anat,norefersub,resource-priority,X-cisco-srtp-fallback,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 217
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 16080 0 IN IP4 172.25.4.26
s=SIP Call
t=0 0
m=audio 28086 RTP/AVP 0 101
c=IN IP4 172.25.4.26
b=TIAS:64000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
030604: .Dec 20 17:43:23.905: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.25.4.200:5060;branch=z9hG4bKB871399
From: "+12025552001" <sip:2001 at 172.25.4.200>;tag=5E6EC01-2547
To: <sip:2004 at 172.25.4.26>;tag=6899cd85aa57010c52826a47-3bf1086e
Call-ID: 3739F3E8-C64411E6-910CD720-9D491D8F at 172.25.4.200
Session-ID:
0ca99dec00105000a0006899cd85aa57;remote=e3d1d5b9cf5d5821bc81fbad4de46749
Date: Tue, 20 Dec 2016 23:41:30 GMT
CSeq: 101 INVITE
Server: Cisco-CP7821/11.0.1
Contact: <sip:445806ff-0c30-04cf-f8ad-3fb72b5119c9 at 172.25.4.26:5060
;transport=udp>;+u.sip!devicename.ccm.cisco.com="SEP6899CD85AA57"
Allow:
ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "Candace Flynn" <sip:2004 at 172.25.4.200
>;party=called;id-type=subscriber;privacy=off;screen=yes
Supported:
replaces,join,sdp-anat,norefersub,resource-priority,X-cisco-srtp-fallback,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 217
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 16080 0 IN IP4 172.25.4.26
s=SIP Call
t=0 0
m=audio 28086 RTP/AVP 0 101
c=IN IP4 172.25.4.26
b=TIAS:64000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
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