Troubleshooting SIP with Cisco Unified
Communications
BRKUCC-2932
Paul Giralt
Distinguished Services Engineer
pgiralt@cisco.com
Agenda
Introduction
Session Initiation Protocol (SIP) Overview
Troubleshooting Tools
Unified CM Tracing
Cisco Unified Border Element (CUBE) Tracing
What is SIP?
Signaling protocol used to establish, manage, and terminate sessions
over an IP network
Core protocol defined in RFC 3261
Extended in many, many other RFCs
ASCII-based messages
What is SIP?
User Agents
SIP Messages
–
Requests and Responses
–
Headers
Media Negotiation
–
Session Description Protocol
–
Offer/Answer Model
–
Early Offer vs. Delayed Offer
–
Early Media
User Agents
User Agent Clients (UAC) send requests to User Agent Servers (UAS)
User Agent Servers send responses to the requests
Most SIP devices are both a UAC and a UAS (they both initiate and accept
requests)
Unified CM and CUBE are both Back-to-Back User Agents (B2BUA) (as
SIP Request Methods from RFC 3261
INVITE
- A user or service is being invited to participate in a multimedia
session
ACK
- Confirms that a client has received a final response to an INVITE
request
BYE
- Terminates an existing session; can be sent by any user agent (in a
multiparty session)
CANCEL
- Cancels pending requests; does not terminate sessions that
have been accepted
OPTIONS
- Queries the capabilities of servers (Also used as a keep alive)
Additional SIP Request Methods
INFO
(RFC 2976) - to send more information within an established dialog
PRACK
(RFC 3262) - to acknowledge a provisional response
SUBSCRIBE
(RFC 3265) - to tell a remote node to look for a certain event
NOTIFY
(RFC 3265) - to respond when that certain event occurs
UPDATE
(RFC 3311) - to update parameters of a session set-up
MESSAGE
(RFC 3428) - SIP instant messaging
REFER
(RFC 3515) – to “refer” one UA to communicate with another UA
PUBLISH
(RFC 3903) - to push UA state information to a compositor/presence
SIP INVITE Method
INVITE sip:+18775551234@172.18.159.231:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK1515b3154665
From: "Test User 1" <sip:9195551111@172.18.106.59>;tag=97903bc0-43adcd-45510543
To: <sip:+18775551234@172.18.159.231>
Call-ID: 7c0ca800-bb01baf9-1468e-3b6a12ac@172.18.106.59
Supported: timer,resource-priority,replaces
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Call-Info: <sip:172.18.106.59:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 2081204224-3137452793-0000000466-0996807340
Session-Expires: 1800
P-Asserted-Identity: "Test User 1" <sip:9195551111@172.18.106.59>
Contact: <sip:9195551111@172.18.106.59:5060>;video;audio
Max-Forwards: 69
Content-Length: 864
SIP Request Line
INVITE sip:+18775551234@172.18.159.231:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK1515b3154665
From: "Test User 1" <sip:9195551111@172.18.106.59>;tag=97903bc0-43adcd-45510543
To: <sip:+18775551234@172.18.159.231>
Call-ID: 7c0ca800-bb01baf9-1468e-3b6a12ac@172.18.106.59
Supported: timer,resource-priority,replaces
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Call-Info: <sip:172.18.106.59:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 2081204224-3137452793-0000000466-0996807340
Session-Expires: 1800
P-Asserted-Identity: "Test User 1" <sip:9195551111@172.18.106.59>
Contact: <sip:9195551111@172.18.106.59:5060>;video;audio
Max-Forwards: 69
Content-Length: 864
Content-Type: application/sdp
SIP Method
SIP Headers
INVITE sip:+18775551234@172.18.159.231:5060 SIP/2.0
Via:
SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK1515b3154665
From:
"Test User 1" <sip:9195551111@172.18.106.59>;tag=97903bc0-43adcd-45510543
To:
<sip:+18775551234@172.18.159.231>
Call-ID:
7c0ca800-bb01baf9-1468e-3b6a12ac@172.18.106.59
Supported:
timer,resource-priority,replaces
User-Agent:
Cisco-CUCM8.6
Allow:
INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER
CSeq:
101 INVITE
Expires:
180
Allow-Events:
presence, kpml
Supported:
X-cisco-srtp-fallback
Supported:
Geolocation
Call-Info:
<sip:172.18.106.59:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid:
2081204224-3137452793-0000000466-0996807340
Session-Expires:
1800
P-Asserted-Identity:
"Test User 1" <sip:9195551111@172.18.106.59>
Contact:
<sip:9195551111@172.18.106.59:5060>;video;audio
Max-Forwards:
69
Content-Length:
864
SIP Response
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bKb5291d44b969a4
From: "Paul Giralt" <sip:89915644@172.18.106.59>;tag=19210123ca7-45568313
To: <sip:+19195551212@10.81.2.30>;tag=253488-726
Date: Mon, 16 Jan 2012 04:00:22 GMT
Call-ID: e59bc600-f1319fa5-b1ea4a-3b6a12ac@172.18.106.59
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.2.T
Reason: Q.850;cause=1
Content-Length: 0
SIP Response
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bKb5291d44b969a4
From: "Paul Giralt" <sip:89915644@172.18.106.59>;tag=19210123ca7-45568313
To: <sip:+19195551212@10.81.2.30>;tag=253488-726
Date: Mon, 16 Jan 2012 04:00:22 GMT
Call-ID: e59bc600-f1319fa5-b1ea4a-3b6a12ac@172.18.106.59
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.2.T
Reason: Q.850;cause=1
Content-Length: 0
Response Code
Free-text Reason
SIP Response
SIP/2.0 404 Not Found
Via
: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bKb5291d44b969a4
From
: "Paul Giralt" <sip:89915644@172.18.106.59>;tag=19210123ca7-45568313
To
: <sip:+19195551212@10.81.2.30>;
tag=253488-726
Date
: Mon, 16 Jan 2012 04:00:22 GMT
Call-ID
: e59bc600-f1319fa5-b1ea4a-3b6a12ac@172.18.106.59
CSeq
: 101 INVITE
Allow-Events
: telephone-event
Server
: Cisco-SIPGateway/IOS-15.2.2.T
Reason
: Q.850;cause=1
Content-Length
: 0
SIP Responses
Response Code
Description
Example
1xx
Informational – Request Received and Continuing to Process
Request
100 Trying
180 Ringing
183 Session Progress
2xx
Success – Action was successfully received, understood, and
accepted
200 OK
202 Acceptable
3xx
Redirection – Another SIP Element needs to be contacted in order
to complete the request
300 Multiple Choices
301 Moved Permanently
302 Moved Temporarily
4xx
Client Error – Request contains bad syntax or cannot be fulfilled at
this server
401 Unauthorized
404 Not Found
406 Not Acceptable
486 Busy Here
488 Not Acceptable Here
5xx
Server Error – Server failed to fulfill an apparently valid request
503 Service Unavailable
6xx
Global Failure – Request is invalid at any server
600 Busy Everywhere
INVITE
200 OK
Session Established
Phone 1
Unified CM
Basic SIP Call Setup
ACK
BYE
200 OK
INVITE
200 OK
Session Established
Phone 1
Unified CM
Basic SIP Call Setup with B2BUA (Unified CM)
ACK
BYE
200 OK
Phone 2
INVITE
200 OK
ACK
BYE
200 OK
CUBESBC (CUBE)
INVITE
200 OK
Phone 1
Basic SIP Call Setup with Unified CM and CUBE
ACK
BYE
200 OK
Unified CM
CUBESBC (CUBE)
INVITE
200 OK
ACK
BYE
200 OK
INVITE
200 OK
ACK
BYE
200 OK
SIP
SP
SBCSP SBC
Session Established
Media Negotiation
SIP leverages the Session Description Protocol (SDP)
(RFC 4566/3266/2327) to communicate media information.
SIP uses the offer/answer model described in RFC 3264 to negotiate media
Offer/Answer Model (RFC 3264)
One endpoint sends an offer SDP containing all the capabilities the endpoint
wishes to negotiate.
SDP contains m lines for each media stream being negotiated (i.e. audio,
video, content channel, etc…)
Receiving endpoint sends an answer SDP that contains the same or a subset
of capabilities received in the offer.
Per RFC 3264, “For each "m=" line in the offer, there MUST be a
corresponding "m=“ line in the answer. The answer MUST contain exactly the
same number of "m=" lines as the offer.”
Session Description Protocol (SDP) - Offer
v=0
o=Cisco-SIPUA 26964 0 IN IP4 172.18.159.152
s=SIP Call
t=0 0
m=audio 29254 RTP/SAVP 0 8 18 102 9 116 124 101
c=IN IP4 172.18.159.152
a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:9 G722/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:124 ISAC/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 25466 RTP/AVP 97
c=IN IP4 172.18.159.152
b=TIAS:1000000
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42801E
a=recvonly
Session Description Protocol (SDP) - Answer
v=0
o=CiscoSystemsCCM-SIP 2000 1 IN IP4 172.18.106.59
s=SIP Call
c=IN IP4 172.18.159.152
t=0 0
m=audio 30308 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 0 RTP/AVP 97
Media Negotiation – Early Offer and Delayed Offer
Initiator of the call can send SDP offer in the INVITE – this is called an Early
Offer (EO)
Receiving endpoint can send the SDP offer in a response if the INVITE did not
contain an offer – this is called a Delayed Offer (DO)
For Early Offer, the answer is sent in a response (usually 200 OK).
INVITE with SDP -
Offer
200 OK with SDP -
Answer
Session Established
Phone 1
Unified CM
Early Offer
ACK (no SDP)
BYE
200 OK
INVITE (no SDP)
200 OK with SDP -
Offer
Session Established
Phone 1
Unified CM
Delayed Offer
ACK with SDP -
Answer
BYE
200 OK
Early Media
Delayed Offer calls do not set up media until the 200 OK (call is answered)
If media is required prior to the call being connected, SIP has provisions for
Early Media
With Early Media on a Delayed Offer call, the offer comes from the terminating
side in a provisional response (e.g. 183 Session Progress)
Originating side sends SDP Answer in a PRACK message (defined in RFC
INVITE (no SDP)
200 OK (INVITE) w/ SDP (should be same as answer)
Session Established
Phone 1
Unified CM
Early Media
ACK
BYE
200 OK
183 Session Progress with SDP -
Offer
PRACK with SDP -
Answer
Media Stream Established
200 OK (PRACK)
Media Re-negotiation
Re-INVITE
Either UA involved in a call can re-INVITE an existing dialog to re-negotiate
parameters for the call.
Cannot re-INVITE until any previous INVITE messages have received a final
response.
Media Re-negotiation
Re-INVITE
INVITE sip:dbe40e44-0dfe-45f1-bd7f-e652098ca344@10.116.101.41:49833;transport=tls SIP/2.0
Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK901f9c72c19221
From: "Paul Giralt" <sip:89915644@172.18.106.59>;tag=15462272~0d0d25d7-4931-4a07-83c6-b82e2c213ca7-45545776
To:
<sip:89915644@172.18.106.59>;
tag=
0022bdd6843100702aae8e5b-4be253be
Date: Wed, 11 Jan 2012 03:08:51 GMT
Call-ID: 8c045780-f0c1fd34-8d838f-3b6a12ac@172.18.106.59
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 104 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Call-Info: <urn:x-cisco-remotecc:callinfo>; security= Authenticated; orientation= from; gci= 2-231448; call-instance= 2
Remote-Party-ID: "Paul Giralt" <sip:89915644@172.18.106.59>;party=calling;screen=yes;privacy=off
Contact: <sip:89915644@172.18.106.59:5061;transport=tls>
Content-Type: application/sdp
Media Re-negotiation
Re-INVITE – Stopping a Media Session
v=0
o=CiscoSystemsCCM-SIP 15462272 2 IN IP4 172.18.106.59
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 19594 RTP/SAVP 9 101
a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX
a=rtpmap:9 G722/8000
a=ptime:20
a=inactive
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 19444 RTP/AVP 126
b=TIAS:1000000
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=42801E;packetization-mode=1;level-asymmetry-allowed=1
a=inactive
a=mid:227796888
Media Re-negotiation
Re-INVITE – Delayed Offer to Re-establish Media Stream
INVITE sip:dbe40e44-0dfe-45f1-bd7f-e652098ca344@10.116.101.41:49833;transport=tls SIP/2.0
Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK901fac34c0fb1b
From: "Paul Giralt" <sip:89915644@172.18.106.59>;tag=15462272~0d0d25d7-4931-4a07-83c6-b82e2c213ca7-45545776
To: <sip:89915644@172.18.106.59>;tag=0022bdd6843100702aae8e5b-4be253be
Date: Wed, 11 Jan 2012 03:08:52 GMT
Call-ID: 8c045780-f0c1fd34-8d838f-3b6a12ac@172.18.106.59
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 106 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Call-Info: <urn:x-cisco-remotecc:callinfo>; security= NotAuthenticated; orientation= from; gci= 2-231448; call-instance= 2
Remote-Party-ID: "Paul Giralt" <sip:89915644@172.18.106.59>;party=calling;screen=yes;privacy=off
Contact: <sip:89915644@172.18.106.59:5061;transport=tls>
Content-Length: 0
Media Re-negotiation
Re-INVITE – Offer in 200 OK
SIP/2.0 200 OK
Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK901fac34c0fb1b
From: "Paul Giralt" <sip:89915644@172.18.106.59>;tag=15462272~0d0d25d7-4931-4a07-83c6-b82e2c213ca7-45545776
To: <sip:89915644@172.18.106.59>;tag=0022bdd6843100702aae8e5b-4be253be
Call-ID: 8c045780-f0c1fd34-8d838f-3b6a12ac@172.18.106.59
Date: Wed, 11 Jan 2012 03:08:52 GMT
CSeq: 106 INVITE
Server: Cisco-CPCIUS/9.2.1
Contact: <sip:dbe40e44-0dfe-45f1-bd7f-e652098ca344@10.116.101.41:49833;transport=tls>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "Paul Giralt" <sip:89915644@172.18.106.59>;party=called;id-type=subscriber;privacy=off;screen=yes
Supported:
replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-5.2.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Recv-Info: conference
Recv-Info: x-cisco-conference
Content-Length: 788
Content-Type: application/sdp
Content-Disposition: session;handling=optional
Media Re-negotiation
Re-INVITE – Offer in 200 OK
v=0
o=Cisco-SIPUA 26259 2 IN IP4 10.116.101.41
s=SIP Call
t=0 0
m=audio 32518 RTP/SAVP 0 8 18 102 9 116 124 101
c=IN IP4 10.116.101.41
a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:102 L16/16000
a=rtpmap:9 G722/8000
a=rtpmap:116 iLBC/8000
a=rtpmap:124 ISAC/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 17614 RTP/AVP 126 97
c=IN IP4 10.116.101.41
b=TIAS:2500000
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=42801F;packetization-mode=1;level-asymmetry-allowed=1
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42801F;packetization-mode=0;level-asymmetry-allowed=1
a=sendrecv
Media Re-negotiation
Re-INVITE – Answer in ACK
ACK sip:dbe40e44-0dfe-45f1-bd7f-e652098ca344@10.116.101.41:49833;transport=tls SIP/2.0
Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK901fb064465a06
From: "Paul Giralt"
<sip:89915644@172.18.106.59>;tag=15462272~0d0d25d7-4931-4a07-83c6-b82e2c213ca7-45545776
To: <sip:89915644@172.18.106.59>;tag=0022bdd6843100702aae8e5b-4be253be
Date: Wed, 11 Jan 2012 03:08:52 GMT
Call-ID: 8c045780-f0c1fd34-8d838f-3b6a12ac@172.18.106.59
Max-Forwards: 70
CSeq: 106 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 446
Media Re-negotiation
Re-INVITE – Answer in ACK – Decline Video Support
v=0
o=CiscoSystemsCCM-SIP 15462272 3 IN IP4 172.18.106.59
s=SIP Call
t=0 0
m=audio 4000 RTP/SAVP 0
c=IN IP4 172.18.106.58
a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendonly
m=video
0
RTP/AVP 126
c=IN IP4 10.116.101.50
b=TIAS:1000000
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=42801E;packetization-mode=1;level-asymmetry-allowed=1
a=mid:227796888
DTMF Relay
3 Methods for passing DTMF digits over a SIP network:
–
RFC 2833
–
SIP NOTIFY
DTMF Relay
RFC 2833
Digits are passed in the RTP stream with a unique payload type
Capability is negotiated in SDP like any other codec
m=audio 30414 RTP/AVP 0 8 116 18 100
101
c=IN IP4 172.18.106.231
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:100 X-NSE/800
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=audio 17236 RTP/AVP 0
101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Offer
Answer
DTMF Relay
SIP NOTIFY
Passes DTMF information in a SIP NOTIFY message telephone-event Event
Negotiated in Call-Info header
INVITE sip:+19195553333@172.18.106.231:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK9843c455840434
From: "Paul Giralt" <sip:9195551234@172.18.106.59>;tag=14902469~0d0d25d7-4931-4a07-83c6
To: <sip:+19195553333@172.18.106.231>
Date: Mon, 13 May 2013 14:48:00 GMT
Call-ID: 1a189580-1901fd20-962c99-3b6a12ac@172.18.106.59
... snip ...
Call-Info: <sip:172.18.106.59:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP
... snip ...
Max-Forwards: 69
Content-Length: 0
Offer
DTMF Relay
SIP NOTIFY
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK9843c455840434
From: "Paul Giralt" <sip:9195551234@172.18.106.59>;tag=14902469~0d0d25d7-4931-4a07-83c6
To: <sip:+19195553333@172.18.106.231>;tag=4363A830-17FC
Call-ID: 1a189580-1901fd20-962c99-3b6a12ac@172.18.106.59
... snip ...
Allow-Events: telephone-event
Call-Info: <sip:172.18.106.231:5060>;method="NOTIFY;Event=telephone-event;Duration=500”
... snip ...
Content-Length: 601
Answer
DTMF Relay
SIP NOTIFY
Digits passed in payload of a NOTIFY message
NOTIFY sip:172.18.106.231:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK98443140152a0a
From: "Paul Giralt" <sip:9195551234@172.18.106.59>;tag=14902469~0d0d25d7-4931-4a07-83c6
To: <sip:+19195553333@172.18.106.231>;tag=4363A830-17FC
Call-ID: 1a189580-1901fd20-962c99-3b6a12ac@172.18.106.59
CSeq: 104 NOTIFY
Max-Forwards: 70
Date: Mon, 13 May 2013 14:48:11 GMT
User-Agent: Cisco-CUCM10.0
Event: telephone-event
Subscription-State: active
Contact: <sip:172.18.106.59:5060>
P-Asserted-Identity: "Paul Giralt" <sip:9195551234@172.18.106.59>
Content-Type: audio/telephone-event
Content-Length: 4
DTMF Relay
SIP KPML
Passes DTMF information in a SIP NOTIFY message kpml Event
Capability advertised in Allow-Events – uses SUBSCRIBE message to subscribe
INVITE sip:+19195554444@172.18.106.231:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK986efd6c4e51e4
From: "Paul Giralt" <sip:9195551234@172.18.106.59>;tag=14918970~0d0d25d7-4931-4a07-83c6
To: <sip:+19195554444@172.18.106.231>
Date: Mon, 13 May 2013 15:05:24 GMT
Call-ID: 885e5780-19110134-96567f-3b6a12ac@172.18.106.59
User-Agent: Cisco-CUCM10.0
... snip ...
Allow-Events: presence, kpml
... snip ...
Session-Expires: 18000
Max-Forwards: 69
Content-Length: 0
Offer
DTMF Relay
SIP KPML
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK986efd6c4e51e4
From: "Paul Giralt" <sip:9195551234@172.18.106.59>;tag=14918970~0d0d25d7-4931-4a07-83c6
To: <sip:+19195554444@172.18.106.231>;tag=437394E8-2E1
Date: Mon, 13 May 2013 15:05:26 GMT
Call-ID: 885e5780-19110134-96567f-3b6a12ac@172.18.106.59
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO
Allow-Events: kpml
, telephone-event
Remote-Party-ID: <sip:9196247285@172.18.106.231>;party=called;screen=no;privacy=off
Contact: <sip:+19196247285@172.18.106.231:5060>
Supported: replaces
Server: Cisco-SIPGateway/IOS-15.2.4.M3
Require: timer
Session-Expires: 18000;refresher=uac
Content-Type: multipart/mixed;boundary=uniqueBoundary
Mime-Version: 1.0
Content-Length: 600
Answer
DTMF Relay
SIP KPML
SUBSCRIBE
sip:9195554444@172.18.106.59:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.106.231:5060;branch=z9hG4bKBAE27139E
From: <sip:+19195551234@172.18.106.231>;tag=437394E8-2E1
To: "Paul Giralt" <sip:9195554444@172.18.106.59>;tag=14918970~0d0d25d7-4931-4a07-83c6
Call-ID: 885e5780-19110134-96567f-3b6a12ac@172.18.106.59
CSeq: 101 SUBSCRIBE
Max-Forwards: 70
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M3
Event: kpml
Expires: 7200
Contact: <sip:172.18.106.231:5060>
Content-Type: application/kpml-request+xml
Content-Length: 327
<?xml version="1.0" encoding="UTF-8"?><
kpml-request
xmlns="urn:ietf:params:xml:ns:kpml-request"
xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
xsi:schemaLocation="urn:ietf:params:xml:ns:kpml-request kpml-xsi:schemaLocation="urn:ietf:params:xml:ns:kpml-request.xsd" version="1.0"><pattern persist="persist">
<regex
tag="dtmf">[x*#ABCD]</regex>
</pattern></kpml-request>
DTMF Relay
SIP KPML
NOTIFY sip:172.18.106.231:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK986f73662cca3b
From: "Paul Giralt" <sip:9195554444@172.18.106.59>;tag=14918970~0d0d25d7-4931-4a07-83c6
To: <sip:+19195551234@172.18.106.231>;tag=437394E8-2E1
Call-ID: 885e5780-19110134-96567f-3b6a12ac@172.18.106.59
CSeq: 104 NOTIFY
Max-Forwards: 70
User-Agent: Cisco-CUCM10.0
Event: kpml
Subscription-State: active;expires=7197
Contact: <sip:9195554444@172.18.106.59:5060>
Content-Type: application/kpml-response+xml
Content-Length: 336
<?xml version="1.0" encoding="UTF-8" ?>
<
kpml-response
xmlns="urn:ietf:params:xml:ns:kpml-response"
xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:schemaLocation="urn:ietf:params:xml:ns:kpml-response kpml-response.xsd" code="200"
digits="1"
forced_flush="false" suppressed="false" tag="dtmf" text="Success" version="1.0"/>
SIP Troubleshooting Tools
Unified CM / SME Tools:
–
Real Time Monitoring Tool / Session Trace
–
TranslatorX
RTMT Session Trace Tool
Session Trace Features
Allows you to search for a call based on calling or called number
Does not depend on Call Detail Records
Session trace only traces SIP sessions in detail
Can display raw SIP messages
Uses correlation tags to include all call legs related to the call selected
On versions 8.5 and 8.6, can only be used on calls for which traces still exist
on the server. Unified CM 9.0 allows viewing traces that have been archived
off-server.
RTMT Session Trace Tool
RTMT Session Trace Tool
TranslatorX Tool
Features
Parses through Unified CM CCM/SDI Trace Files (SDL in 9.0+)
Drag-and-Drop support for .txt as well as .gz files.
Latest version supports IOS CUBE ccsip debugs
Decodes SIP, SCCP, H.323, MGCP, Q.Sig, and ISDN Q.931 messages
Call List based on CDR information in the Traces
Can generate multi-protocol ladder diagrams
Sophisticated filtering capabilities
Download for Windows, Mac OS X, and Linux from:
http://translatorx.cisco.com/
NOTE: Do not call TAC for support on TranslatorX (although many TAC engineers use it so
feel free to mention you’re using it)
TranslatorX Tool
TranslatorX Tool
Call List Filtering
Double-click for complete
Call Detail Record
TranslatorX Tool
TranslatorX Tool
TranslatorX Tool
Call List Filtering
Select a Call and click
“Generate Filter” button
TranslatorX Tool
Wireshark
Open Source network packet capture and analysis tool
Available at
http://www.wireshark.org
Available for Windows, Mac OS X, and UNIX/Linux
Wireshark
Wireshark
Wireshark
How to Gather a Trace?
Both Unified CM and IOS provide a mechanism to gather a packet capture
Unified CM Trace Configuration
SIP messaging in Unified CM is written to the CCM/SDI trace file when
appropriate trace levels are set (SDL trace in 9.0+)
Configured from
Cisco Unified Serviceability
>
Trace
>
Configuration
or by
using AnalysisManager
Unified CM 9.0 combines SDI and SDL traces into the SDL traces
Unified CM 9.0 and later default to detailed tracing – no need to configure
Unified CM Trace Configuration
Select the
Server
Select the Service on
Which Trace Needs to
Be Enabled
Select Service
Group
Unified CM Trace Configuration
1. Press
Set Default
2. Set to
Detailed
Updates All
Servers in This
Cluster with
These Settings
Unified CM Trace Configuration
Enable SIP Stack Trace is NOT needed to see SIP Messages.
Do not enable SIP Stack Trace prior to 9.0 unless directed by TAC
Unified CM Trace Configuration
Can Also Use the
Troubleshooting Trace Settings
Page in
Trace Collection
Various Ways to Collect Trace Files
RTMT Analysis Manager
RTMT Remote Browse
RTMT Collect Files
RTMT Query Wizard
OS CLI (
file get
or
file tail
)
Gathering a Packet Capture from Unified CM
Use the Platform CLI command ‘utils network capture’
admin:utils network capture ? Syntax:
utils network capture [options]
options optional page, numeric, file fname, count num, size bytes, src addr, dest addr, port num, host protocol addr
admin:utils network capture file capturefile count 100000 size ALL host ip 10.1.1.1 Executing command with options:
size=ALL count=100000 interface=eth0 src= dest= port= ip=10.1.1.1
admin:file list activelog platform/cli capturefile.cap dir count = 0, file count = 1
admin:file get activelog platform/cli/capturefile.cap Please wait while the system is gathering files info ...done. Sub-directories were not traversed.
Number of files affected: 1 Total size in Bytes: 24
Total size in Kbytes: 0.0234375 Would you like to proceed [y/n]? y
Cisco Unified Border Element (CUBE) Tracing
Configuration
CUBE Debugging
CUBE / IOS Tools:
–
IOS debugs
–
IOS show commands
–
Per-call trace
CUBE Debugging
When debugging in IOS, configure logging buffered to a fairly large value
(based on available memory)
Disable logging to the console with command ‘no logging console’
Enable timestamps for debugs
Make sure router has NTP enabled
service timestamps debug datetime msec localtime
service timestamps log datetime msec localtime
logging buffered 10000000
no logging console
clock timezone EST -5 0
clock summer-time EDT recurring
ntp server 10.14.1.1
CUBE Debugging
Various SIP debugs available:
CUBE#
debug ccsip ?
all Enable all SIP debugging traces
calls Enable CCSIP SPI calls debugging trace
dhcp Enable SIP-DHCP debugging trace
error Enable SIP error debugging trace
events Enable SIP events debugging trace
function Enable SIP function debugging trace
info Enable SIP info debugging trace
media Enable SIP media debugging trace
messages Enable CCSIP SPI messages debugging trace
preauth Enable SIP preauth debugging traces
states Enable CCSIP SPI states debugging trace
translate Enable SIP translation debugging trace
transport Enable SIP transport debugging traces
verbose Enable verbose mode
CUBE Debugging
Sample ‘debug ccsip messages’
Jan 12 03:14:43.102: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received:
INVITE sip:+19193922900@10.14.1.10:5060 SIP/2.0
Via: SIP/2.0/TCP 172.18.106.59:5060;branch=z9hG4bK978d2e8df73dc
From: "Paul Giralt" <sip:89915644@172.18.106.59>;tag=16218435~-b82e2c213ca7-45552048 To: <sip:+19193922900@10.14.1.10>
Date: Thu, 12 Jan 2012 03:09:42 GMT
Call-ID: ddc4e480-f0e14ef6-94ca5c-3b6a12ac@172.18.106.59 Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY CSeq: 101 INVITE Expires: 180 Allow-Events: presence, kpml Supported: X-cisco-srtp-fallback Supported: Geolocation Call-Info: <sip:172.18.106.59:5060>;method="NOTIFY;Event=telephone-event;Duration=500" Cisco-Guid: 3720668288-0000065536-0000015564-0996807340 Session-Expires: 1800
P-Asserted-Identity: "Paul Giralt" <sip:89915644@172.18.106.59>
Remote-Party-ID: "Paul Giralt" <sip:89915644@172.18.106.59>;party=calling;screen=yes;privacy=off Contact: <sip:89915644@172.18.106.59:5060;transport=tcp>;video;audio
Max-Forwards: 69 Content-Length: 0
CUBE Debugging
Other generic voice debugs can be useful as well:
–
debug voice ccapi inout
–
debug voice dialpeer
–
debug voice rtp session dtmf-relay
Cisco Unified Border Element Basic Call Flow
1.
Incoming VoIP setup message from originating endpoint
2.
This matches inbound VoIP dial peer 1 for characteristics such as codec, VAD, DTMF
method, protocol, etc.
3.
Match the called number to outbound VoIP dial peer 2
4.
Outgoing VoIP setup message
Incoming VoIP Call
Outgoing VoIP Call
dial-peer voice 1 voip destination-pattern 1000 incoming called-number .T session protocol sipv2
session target ipv4:192.168.10.50 dtmf-relay rtp-nte sip-kpml codec g711ulaw
dial-peer voice 2 voip destination-pattern 2000 session protocol sipv2
session target ipv4:192.168.12.25 dtmf-relay rtp-nte codec g711ulaw
Originating
Endpoint
Terminating
Endpoint
CUBEvoice service voip
CUBE
show
Commands
show call active voice [brief]
shows state of currently active calls
0 : 2807 92135710ms.1 (23:55:20.115 EST Mon Jan 16 2012) +1770 pid:1 Answer 89915644 activedur 00:00:14 tx:743/14860 rx:718/14360 dscp:0 media:0 audio tos:0xB8 video tos:0x0
IP 10.116.101.41:23412 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 TextRelay: off media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
0 : 2808 92135720ms.1 (23:55:20.125 EST Mon Jan 16 2012) +1750 pid:100 Originate 9193922900 active
dur 00:00:14 tx:718/14360 rx:755/15100 dscp:0 media:0 audio tos:0xB8 video tos:0x0
IP 172.30.206.164:10076 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 TextRelay: off media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
Telephony call-legs: 0 SIP call-legs: 2 H323 call-legs: 0
Call agent controlled call-legs: 0 SCCP call-legs: 0
Multicast call-legs: 0 Total call-legs: 2
CUBE Per-Call Debugging (PCD)
Useful for CUBE under high call volume
Available on all CUBE(Ent) ASR releases and in 15.1(2)T and later on ISR
All the debug pertaining to a particular call goes into a buffer
“Trigger-points” looks for specific info in the buffers to export the debug info to
an output destination
Can trigger based on user-defined criteria or log every call
–
SIP 4XX, 5XX, or 6XX Response
–
Q.850 Cause code
1. Define buffers and
buffer sizes
2. Turn per-call debugging
on/off
3. Set trigger
points
per-call num-buffer <num>
per-call buffer-size debug <num>
per-call shutdown
per-call active debug
per-call inactive
per-call trigger cause 1
per-call trigger cause 41
per-call trigger sip-message 404
per-call trigger sip-message 488
4. Export debug buffer
content
per-call export primary [flash | ftp | http | pram |
rcp | tftp] secondary [flash | ftp | http | pram | rcp |
tftp]
show per-call stat
show per-call buffer list
router#show per-call buffer content ?
<0-10000000> Specify the buffer num
router#show per-call buffer content 1
6. Show buffer contents on console
5. Show buffer content status
CUBE Per-Call Debugging (PCD)
PCD Configuration
CUBE – IP Traffic Capture
Export Packet Data in PCAP Format
IP Traffic Export feature allows export of packets on an interface
Configuration:
Usage:
ip traffic-export profile CUBE_Debug mode capture
bidirectional
incoming access-list 101
outgoing access-list 101
interface GigabitEthernet0/0
ip traffic-export apply CUBE_Debug size 10000000
traffic-export interface g0/0 start
traffic-export interface g0/0 stop
Case Study 1: Unable to place a call
Problem Description
A user reports that every time they call (919) 555-1212, they get a message that
Case Study 1: Unable to Place a Call
Use RTMT Session Trace
Enter *5551212 into Called Number/URI field
Set time and duration appropriately
Case Study 1: Unable to Place a Call
Use RTMT Session Trace
Double-click to see
message diagram
Clearly shows the
far-end sfar-ends back a 404
Not Found
Case Study 1: Unable to Place a Call
Troubleshoot Call on CUBE
Enable SIP message debugs –
debug ccsip messages
Jan 16 04:00:22.679: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received:
INVITE sip:+19195551212@10.81.2.30:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bKb5291d44b969a4
From: "Paul Giralt" <sip:89915644@172.18.106.59>;tag=19210128~0d0d25d7-4931-4a07-83c6-b82e2c213ca7-45568313 To: <sip:+19195551212@10.81.2.30>
Date: Mon, 16 Jan 2012 03:55:17 GMT
Call-ID: e59bc600-f1319fa5-b1ea4a-3b6a12ac@172.18.106.59 Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY CSeq: 101 INVITE Expires: 180 Allow-Events: presence, kpml Call-Info: <sip:172.18.106.59:5060>;method="NOTIFY;Event=telephone-event;Duration=500" Cisco-Guid: 3852191232-0000065536-0000018595-0996807340 Session-Expires: 1800
P-Asserted-Identity: "Paul Giralt" <sip:89915644@172.18.106.59> Contact: <sip:89915644@172.18.106.59:5060>
Max-Forwards: 69 Content-Length: 0
Case Study 1: Unable to Place a Call
Troubleshoot Call on CUBE
Check to see if the number matches a valid dial peer
Jan 16 04:00:22.687: //98/E59BC6000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bKb5291d44b969a4
From: "Paul Giralt"
<sip:89915644@172.18.106.59>;tag=19210128~0d0d25d7-4931-4a07-83c6-b82e2c213ca7-45568313
To: <sip:+19195551212@10.81.2.30>;tag=253488-726
Date: Mon, 16 Jan 2012 04:00:22 GMT
Call-ID: e59bc600-f1319fa5-b1ea4a-3b6a12ac@172.18.106.59
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.2.T
Reason: Q.850;cause=1
Content-Length: 0
CUBE#
show dialplan number +19195551212
Macro Exp.: +19195551212
Case Study 2: Unable to Place a Call #2
Problem Description
A user reports that every time they call (919) 555-1212, they get reorder (fast
Case Study 2: Unable to Place a Call #2
Use RTMT Session Trace
Enter *5551212 into Called Number/URI field
Set time and duration appropriately
Case Study 2: Unable to Place a Call #2
Use RTMT Session Trace
Trace shows
signaling from phone
as well as to CUBE
CUBE is responding
Case Study 2: Unable to Place a Call #2
Problem Description
As of IOS 15.1(2)T, IOS will reject calls from unknown sources by default
Can either disable the feature or add the list of permitted addresses
voice service voip
no ip address trusted authenticate
allow-connections sip to sip
sip
voice service voip
ip address trusted list
ipv4 172.18.106.0 255.255.255.0
allow-connections sip to sip
sip
OR
Case Study 3: No One Answers the Phone
Problem Description
A user reports that every time they call a specific phone number, no one answers
the call, but if they call from their cell phone, the call is answered immediately
every time.
Calling phone is extension 89919236.
Case Study 3: No One Answers the Phone
Collect Traces
Case Study 3: No One Answers the Phone
Use TranslatorX
Problem is reproducible, so generate a test call and then collect traces. Select
File
>
Open Folder…
Case Study 3: No One Answers the Phone
Use TranslatorX to Analyze Traces
Case Study 3: No One Answers the Phone
Use TranslatorX to Analyze Traces
Disable Filters
Select the INVITE
Case Study 3: No One Answers the Phone
Use TranslatorX to Analyze Traces
03/29/2010 10:36:41.497 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 172.18.159.231:[5060]:
INVITE sip:+18772888362@172.18.159.231:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK1515b3154665
From: "Test User 1" <sip:9194769236@172.18.106.59>;tag=97903bc0-a3de-4a15-ba27-44c81fe3adcd-45510543
To: <sip:+18772888362@172.18.159.231>
Date: Mon, 29 Mar 2010 14:36:41 GMT
Call-ID: 7c0ca800-bb01baf9-1468e-3b6a12ac@172.18.106.59
Supported: timer,resource-priority,replaces Min-SE: 1800
User-Agent: Cisco-CUCM8.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY CSeq: 101 INVITE Expires: 180 Allow-Events: presence, kpml Supported: X-cisco-srtp-fallback Supported: Geolocation Call-Info: <sip:172.18.106.59:5060>;method="NOTIFY;Event=telephone-event;Duration=500" Cisco-Guid: 2081204224-3137452793-0000000466-0996807340 Session-Expires: 1800
P-Asserted-Identity: "Test User 1" <sip:9194769236@172.18.106.59> Contact: <sip:9194769236@172.18.106.59:5060>;video;audio Max-Forwards: 69
Case Study 3: No One Answers the Phone
Use TranslatorX to Analyze Traces
Case Study 3: No One Answers the Phone
Use TranslatorX to Analyze Traces
Select the INVITE
Create New Filter (control/command-N)
Filter by IP Address (control/command – I)
Case Study 3: No One Answers the Phone
Case Study 3: No One Answers the Phone
INVITE from IP Phone w/ SDP
03/29/2010 10:36:33.771 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 172.18.159.152 on port 51682 index 2321
with 1717 bytes:
INVITE sip:9@172.18.106.59;user=phone SIP/2.0
Via: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK1636ab61
From: "Test User 1" <sip:89919236@172.18.106.59>
;tag=00260bd9669e07147bcb3aac-3cda8f0c
To: <sip:9@172.18.106.59;user=phone>
Call-ID: 00260bd9-669e000b-588c0c2b-2193e2a3@172.18.159.152
Max-Forwards: 70
Date: Mon, 29 Mar 2010 14:36:33 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP9951/9.0.1
Contact: <sip:4a8a8f91-609e-d655-19ea-44eedcd7b0d6@172.18.159.152:51682;transport=tls>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "Test User 1" <sip:89919236@172.18.106.59>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported:
replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-5.0.0,X-cisco-xsi-9.0.1
Allow-Events: kpml,dialog
Content-Length: 632
Content-Type: application/sdp
Content-Disposition: session;handling=optional
Case Study 3: No One Answers the Phone
INVITE from IP Phone w/ SDP (continued)
v=0 o=Cisco-SIPUA 26964 0 IN IP4 172.18.159.152 s=SIP Call t=0 0 m=audio 29254 RTP/SAVP 0 8 18 102 9 116 124 101 c=IN IP4 172.18.159.152 a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:102 L16/16000 a=rtpmap:9 G722/8000 a=rtpmap:116 iLBC/8000 a=fmtp:116 mode=20 a=rtpmap:124 ISAC/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv m=video 25466 RTP/AVP 97 c=IN IP4 172.18.159.152 b=TIAS:1000000 a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=42801E a=recvonly
Case Study 3: No One Answers the Phone
Unified CM Sends a 100 Trying
03/29/2010 10:36:33.773 |//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 172.18.159.152 on port 51682 index 2321
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK1636ab61
From: "Test User 1" <sip:89919236@172.18.106.59>;tag=00260bd9669e07147bcb3aac-3cda8f0c
To: <sip:9@172.18.106.59;user=phone>
Date: Mon, 29 Mar 2010 14:36:33 GMT
Call-ID: 00260bd9-669e000b-588c0c2b-2193e2a3@172.18.159.152
CSeq: 101 INVITE Allow-Events: presence Content-Length: 0
Case Study 3: No One Answers the Phone
Unified CM Sends a REFER to Play Outside Dialtone
03/29/2010 10:36:33.780 |//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 172.18.159.152 on port 51682 index 2321
REFER sip:89919236@172.18.159.152:51682 SIP/2.0
Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK151511c5f04bf
From: <sip:89919236@172.18.106.59>;tag=2144536187 To: <sip:89919236@172.18.159.152> Call-ID: 7747f400-bb01baf1-14685-3b6a12ac@172.18.106.59 CSeq: 101 REFER Max-Forwards: 70 Contact: <sip:89919236@172.18.106.59:5061;transport=tls> User-Agent: Cisco-CUCM8.0 Expires: 0 Refer-To: cid:1234567890@172.18.106.59 Content-Id: <1234567890@172.18.106.59> Require: norefersub Content-Type: application/x-cisco-remotecc-request+xml Referred-By: <sip:89919236@172.18.106.59> Content-Length: 409
Case Study 3: No One Answers the Phone
Unified CM Sends a REFER to play Outside Dialtone (continued)
<x-cisco-remotecc-request> <playtonereq> <dialogid> <callid>00260bd9-669e000b-588c0c2b-2193e2a3@172.18.159.152</callid> <localtag>97903bc0-a3de-4a15-ba27-44c81fe3adcd-45510542</localtag> <remotetag>00260bd9669e07147bcb3aac-3cda8f0c</remotetag> </dialogid>
<tonetype>DtOutsideDialTone</tonetype> <direction>user</direction>
</playtonereq>
Case Study 3: No One Answers the Phone
Unified CM Sends a SUBSCRIBE for KPML
03/29/2010 10:36:33.781 |//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 172.18.159.152 on port 51682 index 2321
SUBSCRIBE sip:89919236@172.18.159.152:51682 SIP/2.0
Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK1515232b4e84f
From: <sip:9@172.18.106.59>;tag=1976165806
To: <sip:89919236@172.18.159.152>
Call-ID: 7747f400-bb01baf1-14686-3b6a12ac@172.18.106.59
CSeq: 101 SUBSCRIBE
Date: Mon, 29 Mar 2010 14:36:33 GMT User-Agent: Cisco-CUCM8.0
Event: kpml; call-id=00260bd9-669e000b-588c0c2b-2193e2a3@172.18.159.152; from-tag=00260bd9669e07147bcb3aac-3cda8f0c Expires: 7200
Contact: <sip:9@172.18.106.59:5061;transport=tls> Accept: application/kpml-response+xml
Max-Forwards: 70
Content-Type: application/kpml-request+xml
Content-Length: 424
<?xml version="1.0" encoding="UTF-8" ?>
<kpml-request xmlns="urn:ietf:params:xml:ns:kpml-request" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:schemaLocation="urn:ietf:params:xml:ns:kpml-request kpml-request.xsd" version="1.0">
<pattern criticaldigittimer="1000" extradigittimer="500" interdigittimer="10000" persist="persist"> <regex tag="Backspace OK">[x#*+]|bs</regex>
</pattern> </kpml-request>
Case Study 3: No One Answers the Phone
Phone Sends 200 OK for the REFER and SUBSCRIBE
03/29/2010 10:36:33.802 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 172.18.159.152 on port 51682 index 2321 with 453 bytes:
SIP/2.0 200 OK
Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK151511c5f04bf
From: <sip:89919236@172.18.106.59>;tag=2144536187
To: <sip:89919236@172.18.159.152>;tag=00260bd9669e07167c743311-343ee3af
Call-ID: 7747f400-bb01baf1-14685-3b6a12ac@172.18.106.59
Date: Mon, 29 Mar 2010 14:36:33 GMT CSeq: 101 REFER
Server: Cisco-CP9951/9.0.1
Contact: <sip:4a8a8f91-609e-d655-19ea-44eedcd7b0d6@172.18.159.152:51682;transport=TLS> Content-Length: 0
03/29/2010 10:36:33.843 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 172.18.159.152 on port 51682 index 2321 with 465 bytes:
SIP/2.0 200 OK
Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK1515232b4e84f
From: <sip:9@172.18.106.59>;tag=1976165806
To: <sip:89919236@172.18.159.152>;tag=00260bd9669e07177ee0d51d-14f56f89
Call-ID: 7747f400-bb01baf1-14686-3b6a12ac@172.18.106.59
Date: Mon, 29 Mar 2010 14:36:33 GMT CSeq: 101 SUBSCRIBE
Server: Cisco-CP9951/9.0.1
Contact: <sip:4a8a8f91-609e-d655-19ea-44eedcd7b0d6@172.18.159.152:51682;transport=TLS> Expires: 7200
Case Study 3: No One Answers the Phone
IP Phone
(172.18.159.152)Unified CM
(172.18.159.152)CUBE
(172.18.159.231) INVITE 100 Trying REFER SUBSCRIBE 200 OK (REFER) 200 OK (SUBSCRIBE)Case Study 3: No One Answers the Phone
User Dials a ‘1’
03/29/2010 10:36:34.350 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 172.18.159.152 on port 51682 index 2321 with 896 bytes:
NOTIFY sip:9@172.18.106.59:5061 SIP/2.0
Via: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK1cd529ba
To: <sip:9@172.18.106.59>;tag=1976165806
From: <sip:89919236@172.18.159.152>;tag=00260bd9669e07177ee0d51d-14f56f89
Call-ID: 7747f400-bb01baf1-14686-3b6a12ac@172.18.106.59
Date: Mon, 29 Mar 2010 14:36:33 GMT CSeq: 1001 NOTIFY
Event: kpml
Subscription-State: active; expires=7200 Max-Forwards: 70 Contact: <sip:4a8a8f91-609e-d655-19ea-44eedcd7b0d6@172.18.159.152:51682;transport=TLS> Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Content-Length: 209 Content-Type: application/kpml-response+xml Content-Disposition: session;handling=required <?xml version="1.0" encoding="UTF-8"?>
<kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" version="1.0" code="200" text="OK" suppressed="false" forced_flush="false"
Case Study 3: No One Answers the Phone
Unified CM Replies to NOTIFY With a 200 OK
03/29/2010 10:36:34.352 |//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 172.18.159.152 on port 51682 index 2321
SIP/2.0 200 OK
Via: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK1cd529ba
From: <sip:89919236@172.18.159.152>;tag=00260bd9669e07177ee0d51d-14f56f89 To: <sip:9@172.18.106.59>;tag=1976165806
Date: Mon, 29 Mar 2010 14:36:34 GMT
Call-ID: 7747f400-bb01baf1-14686-3b6a12ac@172.18.106.59
CSeq: 1001 NOTIFY Content-Length: 0
Case Study 3: No One Answers the Phone
Unified CM Replies Sends a REFER to Disable Outside Dialtone
03/29/2010 10:36:34.353 |//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 172.18.159.152 on port 51682 index 2321
REFER sip:89919236@172.18.159.152:51682 SIP/2.0
Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK151536ea86ab0
From: <sip:89919236@172.18.106.59>;tag=1574166193 To: <sip:89919236@172.18.159.152> Call-ID: 77e08a80-bb01baf2-14687-3b6a12ac@172.18.106.59 CSeq: 101 REFER Max-Forwards: 70 Contact: <sip:89919236@172.18.106.59:5061;transport=tls> User-Agent: Cisco-CUCM8.0 Expires: 0 Refer-To: cid:1234567890@172.18.106.59 Content-Id: <1234567890@172.18.106.59> Require: norefersub
Content-Type: application/x-cisco-remotecc-request+xml
Referred-By: <sip:89919236@172.18.106.59> Content-Length: 401
Case Study 3: No One Answers the Phone
<x-cisco-remotecc-request> <playtonereq>
<dialogid>
<callid>00260bd9-669e000b-588c0c2b-2193e2a3@172.18.159.152</callid> <localtag>97903bc0-a3de-4a15-ba27-44c81fe3adcd-45510542</localtag> <remotetag>00260bd9669e07147bcb3aac-3cda8f0c</remotetag>
</dialogid>
<tonetype>Dt_NoTone</tonetype> <direction>user</direction>
</playtonereq>
Case Study 3: No One Answers the Phone
Phone Replies With 200 OK to REFER
03/29/2010 10:36:34.402 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 172.18.159.152 on port 51682 index 2321 with 453 bytes:
SIP/2.0 200 OK
Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK151536ea86ab0
From: <sip:89919236@172.18.106.59>;tag=1574166193
To: <sip:89919236@172.18.159.152>;tag=00260bd9669e07184b08b96b-796ab86f
Call-ID: 77e08a80-bb01baf2-14687-3b6a12ac@172.18.106.59
Date: Mon, 29 Mar 2010 14:36:33 GMT CSeq: 101 REFER
Server: Cisco-CP9951/9.0.1
Contact: <sip:4a8a8f91-609e-d655-19ea-44eedcd7b0d6@172.18.159.152:51682;transport=TLS> Content-Length: 0