07-23-2015 12:43 AM - edited 03-17-2019 03:44 AM
Hi,
if a call comes from itsp, 15 secound after the first invite cube sends a CANCEL with Reason: Q.850;cause=102 to CUCM and SIP/2.0 408 Request Timeout with Reason: Q.850;cause=102
I can not find any timer that is 15 secounds on the cube:
isr2#sh sip-ua timers
SIP UA Timer Values (millisecs unless noted)
trying 500, expires 180000, connect 500, disconnect 500
prack 500, rel1xx 500, notify 200, update 500
refer 500, register 500, info 500, options 500, hold 2880 minutes
, registrar-dns-cache 3600 seconds
tcp/udp aging 5 minutes
tls aging 60 minutes
isr2#
Jul 22 15:22:10.082 LUEBECK: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:+4945131012012@*.*.*.4:5061;user=phone;transport=tls SIP/2.0
Via: SIP/2.0/TLS *.*.*.8:5062;branch=z9hG4bK2i3s6h20cg21vs0qa460.1
From: <sip:+494515005413@*.*.*.8:54390;user=phone>;tag=gK0c771273
To: <sip:+4945131012012@*.*.*.4:5061;user=phone>
Call-ID: 439106021_56042896@10.59.69.122
CSeq: 354168095 INVITE
Max-Forwards: 55
Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+494515005413@*.*.*.8:5062;transport=tls>
Supported: timer,100rel,histinfo,precondition,replaces
History-Info: <sip:unknown@unknown.invalid;cause=404>;index=1.1, <sip:+4945131012012@tel.t-online.de;user=phone>;index=1.1.1;np=1.1, <sip:+4945131012012@tg1001.dfn.siptg.t-systems.com;user=phone;target=+4945131012012%40tel.t-online.de>;index=1.1.1.1;rc=1.1.1
Session-Expires: 120
Min-SE: 90
Content-Length: 438
Content-Disposition: session; handling=required
Content-Type: application/sdp
v=0
o=Sonus_UAC 1225232477 740026466 IN IP4 *.*.*.9
s=SIP Media Capabilities
c=IN IP4 *.*.*.9
t=0 0
m=audio 39610 RTP/SAVP 9 8 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ZZxL7AB8U9h9nK6sVpuYMKrfelUtVO55cV44LVkU
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:gRHgGmBIIY1tunyHpE9EEKm+XcNzrVxv5dj/uA01
Jul 22 15:22:10.086 LUEBECK: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 422 Session Timer too small
Via: SIP/2.0/TLS *.*.*.8:5062;branch=z9hG4bK2i3s6h20cg21vs0qa460.1
From: <sip:+494515005413@*.*.*.8:54390;user=phone>;tag=gK0c771273
To: <sip:+4945131012012@*.*.*.4:5061;user=phone>;tag=42D2DA3C-1D6
Date: Wed, 22 Jul 2015 13:22:10 GMT
Call-ID: 439106021_56042896@10.59.69.122
CSeq: 354168095 INVITE
Allow-Events: telephone-event
Min-SE: 1800
Server: Cisco-SIPGateway/IOS-15.4.3.M3
Content-Length: 0
Jul 22 15:22:10.102 LUEBECK: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:+4945131012012@*.*.*.4:5061;user=phone;transport=tls SIP/2.0
Via: SIP/2.0/TLS *.*.*.8:5062;branch=z9hG4bK2i3s6h20cg21vs0qa460.1
CSeq: 354168095 ACK
From: <sip:+494515005413@*.*.*.8:54390;user=phone>;tag=gK0c771273
To: <sip:+4945131012012@*.*.*.4:5061;user=phone>;tag=42D2DA3C-1D6
Call-ID: 439106021_56042896@10.59.69.122
Max-Forwards: 55
Content-Length: 0
Jul 22 15:22:10.122 LUEBECK: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:+4945131012012@*.*.*.4:5061;user=phone;transport=tls SIP/2.0
Via: SIP/2.0/TLS *.*.*.8:5062;branch=z9hG4bK22av8q20doaghsoe1730.1
From: <sip:+494515005413@*.*.*.8:54390;user=phone>;tag=gK0c771273
To: <sip:+4945131012012@*.*.*.4:5061;user=phone>
Call-ID: 439106021_56042896@10.59.69.122
CSeq: 354168096 INVITE
Max-Forwards: 55
Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <sip:+494515005413@*.*.*.8:5062;transport=tls>
Supported: timer,100rel,histinfo,precondition,replaces
History-Info: <sip:unknown@unknown.invalid;cause=404>;index=1.1, <sip:+4945131012012@tel.t-online.de;user=phone>;index=1.1.1;np=1.1, <sip:+4945131012012@tg1001.dfn.siptg.t-systems.com;user=phone;target=+4945131012012%40tel.t-online.de>;index=1.1.1.1;rc=1.1.1
Session-Expires: 1800
Min-SE: 1800
Content-Length: 438
Content-Disposition: session; handling=required
Content-Type: application/sdp
v=0
o=Sonus_UAC 1225232477 740026466 IN IP4 *.*.*.9
s=SIP Media Capabilities
c=IN IP4 *.*.*.9
t=0 0
m=audio 39442 RTP/SAVP 9 8 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:kLDhTYXo+lV48vXMW6HyJoYwOT6oZIdibIWn58BT
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:CrlhD1fBEGDPPjFemVXt81b0u1yaWal4+lw5gIL4
Jul 22 15:22:10.130 LUEBECK: //138999/7DCDE9BBBD3E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TLS *.*.*.8:5062;branch=z9hG4bK22av8q20doaghsoe1730.1
From: <sip:+494515005413@*.*.*.8:54390;user=phone>;tag=gK0c771273
To: <sip:+4945131012012@*.*.*.4:5061;user=phone>
Date: Wed, 22 Jul 2015 13:22:10 GMT
Call-ID: 439106021_56042896@10.59.69.122
CSeq: 354168096 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.4.3.M3
Content-Length: 0
Jul 22 15:22:10.138 LUEBECK: //139000/7DCDE9BBBD3E/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:+4945131012012@*.*.*.101:5061 SIP/2.0
Via: SIP/2.0/TLS *.*.*.4:5061;branch=z9hG4bKEFE586
From: <sip:+494515005413@*.*.*.4>;tag=42D2DA6C-F4
To: <sip:+4945131012012@*.*.*.101>
Date: Wed, 22 Jul 2015 13:22:10 GMT
Call-ID: 7DCFBE33-2FAB11E5-BD44CD80-56D2F9A3@*.*.*.4
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2110646715-0799740389-3175009664-1456667043
User-Agent: Cisco-SIPGateway/IOS-15.4.3.M3
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1437571330
Contact: <sip:+494515005413@*.*.*.4:5061;transport=tls>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 54
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 318
v=0
o=CiscoSystemsSIP-GW-UserAgent 5758 3053 IN IP4 *.*.*.4
s=SIP Call
c=IN IP4 *.*.*.4
t=0 0
m=audio 26154 RTP/SAVP 9 8
c=IN IP4 *.*.*.4
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:kLDhTYXo+lV48vXMW6HyJoYwOT6oZIdibIWn58BT
a=rtpmap:9 G722/8000
a=fmtp:9 bitrate=64
a=rtpmap:8 PCMA/8000
a=ptime:20
Jul 22 15:22:10.142 LUEBECK: //139000/7DCDE9BBBD3E/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/TLS *.*.*.4:5061;branch=z9hG4bKEFE586
From: <sip:+494515005413@*.*.*.4>;tag=42D2DA6C-F4
To: <sip:+4945131012012@*.*.*.101>
Date: Wed, 22 Jul 2015 13:22:10 GMT
Call-ID: 7DCFBE33-2FAB11E5-BD44CD80-56D2F9A3@*.*.*.4
CSeq: 101 INVITE
Allow-Events: presence
Content-Length: 0
Jul 22 15:22:10.194 LUEBECK: //139000/7DCDE9BBBD3E/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS *.*.*.4:5061;branch=z9hG4bKEFE586
From: <sip:+494515005413@*.*.*.4>;tag=42D2DA6C-F4
To: <sip:+4945131012012@*.*.*.101>;tag=78971~4498ec3c-29e8-4cb2-abe3-b65eefebc6a9-43194542
Date: Wed, 22 Jul 2015 13:22:10 GMT
Call-ID: 7DCFBE33-2FAB11E5-BD44CD80-56D2F9A3@*.*.*.4
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence
Server: Cisco-CUCM10.5
Call-Info: <urn:x-cisco-remotecc:callinfo>; security= Unknown; gci= 2-7019; isVoip
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP
Supported: X-cisco-srtp-fallback
Supported: Geolocation
P-Asserted-Identity: <sip:+4945131012012@*.*.*.101>
Remote-Party-ID: <sip:+4945131012012@*.*.*.101>;party=called;screen=yes;privacy=off
Contact: <sip:+4945131012012@*.*.*.101:5061;transport=tls>
Content-Length: 0
Jul 22 15:22:10.194 LUEBECK: //138999/7DCDE9BBBD3E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS *.*.*.8:5062;branch=z9hG4bK22av8q20doaghsoe1730.1
From: <sip:+494515005413@*.*.*.8:54390;user=phone>;tag=gK0c771273
To: <sip:+4945131012012@*.*.*.4:5061;user=phone>;tag=42D2DAA8-2482
Date: Wed, 22 Jul 2015 13:22:10 GMT
Call-ID: 439106021_56042896@10.59.69.122
CSeq: 354168096 INVITE
Require: 100rel
RSeq: 3044
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:+4945131012012@*.*.*.4:5061;transport=tls>
Server: Cisco-SIPGateway/IOS-15.4.3.M3
Content-Length: 0
Jul 22 15:22:10.214 LUEBECK: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
PRACK sip:+4945131012012@*.*.*.4:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS *.*.*.8:5062;branch=z9hG4bK32n3dc30e0khvs0ns5s0.1
From: <sip:+494515005413@*.*.*.8:54390;user=phone>;tag=gK0c771273
To: <sip:+4945131012012@*.*.*.4:5061;user=phone>;tag=42D2DAA8-2482
Call-ID: 439106021_56042896@10.59.69.122
CSeq: 354168097 PRACK
Max-Forwards: 69
RAck: 3044 354168096 INVITE
Content-Length: 0
Jul 22 15:22:10.214 LUEBECK: //138999/7DCDE9BBBD3E/SIP/Msg/ccsipDisplayMsg:
Sent:
isr2#SIP/2.0 200 OK
Via: SIP/2.0/TLS *.*.*.8:5062;branch=z9hG4bK32n3dc30e0khvs0ns5s0.1
From: <sip:+494515005413@*.*.*.8:54390;user=phone>;tag=gK0c771273
To: <sip:+4945131012012@*.*.*.4:5061;user=phone>;tag=42D2DAA8-2482
Date: Wed, 22 Jul 2015 13:22:10 GMT
Call-ID: 439106021_56042896@10.59.69.122
Server: Cisco-SIPGateway/IOS-15.4.3.M3
CSeq: 354168097 PRACK
Content-Length: 0
isr2#
Jul 22 15:22:25.134 LUEBECK: //139000/7DCDE9BBBD3E/SIP/Msg/ccsipDisplayMsg:
Sent:
CANCEL sip:+4945131012012@*.*.*.101:5061 SIP/2.0
Via: SIP/2.0/TLS *.*.*.4:5061;branch=z9hG4bKEFE586
From: <sip:+494515005413@*.*.*.4>;tag=42D2DA6C-F4
To: <sip:+4945131012012@*.*.*.101>
Date: Wed, 22 Jul 2015 13:22:10 GMT
Call-ID: 7DCFBE33-2FAB11E5-BD44CD80-56D2F9A3@*.*.*.4
CSeq: 101 CANCEL
Max-Forwards: 70
Timestamp: 1437571345
Reason: Q.850;cause=102
Content-Length: 0
Jul 22 15:22:25.134 LUEBECK: //138999/7DCDE9BBBD3E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 408 Request Timeout
Via: SIP/2.0/TLS *.*.*.8:5062;branch=z9hG4bK22av8q20doaghsoe1730.1
From: <sip:+494515005413@*.*.*.8:54390;user=phone>;tag=gK0c771273
To: <sip:+4945131012012@*.*.*.4:5061;user=phone>;tag=42D2DAA8-2482
Date: Wed, 22 Jul 2015 13:22:10 GMT
Call-ID: 439106021_56042896@10.59.69.122
CSeq: 354168096 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.4.3.M3
Reason: Q.850;cause=102
Content-Length: 0
I hope someone has an idea, please let me know if you need more information.
Paul
07-23-2015 06:09 AM
OK I found it.
Debugging told me:
ul 23 14:30:08.111 LUEBECK: //152736/5A64DB51B328/SIP/Info/notify/8193/sipSPIRtpDiscTimerExpired: RTP/RTCP receive timer expired. Disconnect the call.
This page :
http://www.cisco.com/c/en/us/td/docs/ios/12_2/12_2x/12_2xb/feature/guide/ftsiprtp.html#wp1039389
-told me that:
ip rtcp report interval 3000 * timer receive-rtcp 5 = 15 secounds
I increased ip rtcp report interval 3000 to 6000 so I got a Cancel after 30 secounds with no RTCP.
Thx 4 reading.
07-23-2015 06:24 AM
Always great to read about new issues and resolutions (+5)
01-16-2017 08:26 PM
i am experiencing the same issue but randomly. call starts and then drops after about 15 seconds. when my sip provider send me their debugs they see my cube internal ip instead of the nat ip. seems the asa inspection sometimes works and sometimes does not work.
during those 15 seconds the 2 parties can hear each other fine.
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