#CiscoLive
Troubleshooting SIP with Cisco
Unified Communications
Paul Giralt, Distinguished Engineer
pgiralt@cisco.com - @PaulGiralt
BRKUCC-2932
#CiscoLive
Agenda
Agenda
•
Session Initiation Protocol (SIP) Overview
•
Troubleshooting Tools
•
Unified CM Tracing
•
Expressway / VCS Tracing
•
Cisco Unified Border Element (CUBE)
Tracing
•
Sample Call Flows / Case Studies
•
Live Demos
#CiscoLive
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3
SIP Protocol Overview
What is SIP?
•
Signaling protocol used to establish, manage, and terminate sessions over
an IP network
•
Core protocol defined in RFC 3261
•
Relies heavily on RFC 3262, RFC 3263, RFC 3264, and RFC 3265
•
Extended in many, many other RFCs
•
ASCII-based messages
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5
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
DTMF Relay
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6
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 opposed to Proxies)
•
Cisco VCS and Cisco Expressway operate as both proxies and B2BUA’s
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7
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)
•
REGISTER - Registers the user agent with the registrar server of a domain
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8
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 server
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9
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-CUCM12.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Session-ID: 52c41df700105000a00074a02fc0cf3b;remote=00000000000000000000000000000000
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
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10
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
Request URI
SIP Version
Supported: timer,resource-priority,replaces
User-Agent: Cisco-CUCM12.0
Allow: INVITE,
SIP OPTIONS,
MethodINFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Session-ID: 52c41df700105000a00074a02fc0cf3b;remote=00000000000000000000000000000000
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
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11
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-CUCM12.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Session-ID: 52c41df700105000a00074a02fc0cf3b;remote=00000000000000000000000000000000
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
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12
SIP Response
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bKb5291d44b969a4
From: "Test User 1" <sip:9195551111@172.18.106.59>;tag=97903bc0-43adcd-45510543
To: <sip:+18775551234@172.18.159.231>;tag=253488-726
Date: Mon, 16 Jan 2018 04:00:22 GMT
Call-ID: 7c0ca800-bb01baf9-1468e-3b6a12ac@172.18.106.59
CSeq: 101 INVITE
Allow-Events: telephone-event
Session-ID: 747a0ead00105000a00074a02fc0cf3b;remote=52c41df700105000a00074a02fc0cf3b
Server: Cisco-SIPGateway/IOS-15.6.2.T
Reason: Q.850;cause=1
Content-Length: 0
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13
SIP Response
Free-text Reason
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bKb5291d44b969a4
From: "Test User 1" <sip:9195551111@172.18.106.59>;tag=97903bc0-43adcd-45510543
To: <sip:+18775551234@172.18.159.231>;tag=253488-726
Response
Code GMT
Date: Mon, 16 Jan
2012 04:00:22
Call-ID: 7c0ca800-bb01baf9-1468e-3b6a12ac@172.18.106.59
CSeq: 101 INVITE
Allow-Events: telephone-event
Session-ID: 747a0ead00105000a00074a02fc0cf3b;remote=52c41df700105000a00074a02fc0cf3b
Server: Cisco-SIPGateway/IOS-15.6.2.T
Reason: Q.850;cause=1
Content-Length: 0
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14
SIP Response
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bKb5291d44b969a4
From: "Test User 1" <sip:9195551111@172.18.106.59>;tag=97903bc0-43adcd-45510543
To: <sip:+18775551234@172.18.159.231>;tag=253488-726
Date: Mon, 16 Jan 2018 04:00:22 GMT
Call-ID: 7c0ca800-bb01baf9-1468e-3b6a12ac@172.18.106.59
CSeq: 101 INVITE
Allow-Events: telephone-event
Session-ID: 747a0ead00105000a00074a02fc0cf3b;remote=52c41df700105000a00074a02fc0cf3b
Server: Cisco-SIPGateway/IOS-15.6.2.T
Reason: Q.850;cause=1
Content-Length: 0
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15
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
404
406
486
488
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
603 Decline
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Unauthorized
Not Found
Not Acceptable
Busy Here
Not Acceptable Here
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16
SIP Mandatory Headers – RFC 3261 Section 20
• Where column:
• R: header field may only appear in requests;
• r: header field may only appear in responses;
• 2xx, 4xx, etc.: A numerical value or range indicates
response codes with which the header field can be used;
• c: header field is copied from the request to the response.
• Requirements:
• c: Conditional; requirements on the header field depend
on the context of the message.
• m: The header field is mandatory.
• m*: The header field SHOULD be sent, but clients/servers
need to be prepared to receive messages without that
header field.
• o: The header field is optional.
• t: The header field SHOULD be sent, but clients/servers
need to be prepared to receive messages without that
header field. If a stream-based protocol (such as TCP) is
used as a transport, then the header field MUST be sent.
• *: The header field is required if the message body is not
empty. See Sections 20.14, 20.15 and 7.4 for details.
#CiscoLive
Header field
where
proxy ACK BYE CAN INV OPT REG
___________________________________________________________
Accept
R
o
o
m* o
Accept
2xx
o
m* o
Accept
415
c
c
c
c
Accept-Encoding
R
o
o
o
o
Accept-Encoding
2xx
o
m* o
Accept-Encoding
415
c
c
c
c
Accept-Language
R
o
o
o
o
Accept-Language
2xx
o
m* o
Accept-Language
415
c
c
c
c
Alert-Info
R
ar
o
Alert-Info
180
ar
o
Allow
R
o
o
o
o
Allow
2xx
o
m* m* o
Allow
r
o
o
o
o
Allow
405
m
m
m
m
Authentication-Info
2xx
o
o
o
o
Authorization
R
o
o
o
o
o
o
Call-ID
c
r
m
m
m
m
m
m
Call-Info
ar
o
o
o
Contact
R
o
m
o
o
Contact
1xx
o
Contact
2xx
m
o
o
Contact
3xx
d
o
o
o
o
Contact
485
o
o
o
o
Content-Disposition
o
o
o
o
o
Content-Encoding
o
o
o
o
o
Content-Language
o
o
o
o
o
Content-Length
ar
t
t
t
t
t
t
Content-Type
*
*
*
*
*
CSeq
c
r
m
m
m
m
m
m
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17
Basic SIP Call Setup
Unified CM
Phone
INVITE
200 OK
ACK
Session Established
BYE
200 OK
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18
Basic SIP Call Setup with B2BUA
Phone 1
SBC
(CUBE)
Phone
2
Unified CM
INVITE
INVITE
200 OK
200 OK
ACK
ACK
Session Established
BYE
BYE
200 OK
200 OK
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19
Basic SIP Call Setup with UCM and CUBE
Phone
Unified CM
SP SBC
SBC (CUBE)
SBC
INVITE
200 OK
ACK
INVITE
SIP
SP
INVITE
200 OK
200 OK
ACK
ACK
Session Established
BYE
200 OK
BYE
200 OK
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BYE
200 OK
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20
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 using SDP
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21
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.”
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22
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
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23
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
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24
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).
•
For Delayed Offer, the answer is typically sent in the ACK.
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25
Early Offer
Phone
Unified CM
INVITE with SDP - Offer
200 OK with SDP - Answer
ACK (no SDP)
Session Established
BYE
200 OK
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26
Delayed Offer
Phone
Unified CM
INVITE (no SDP)
200 OK with SDP - Offer
ACK with SDP - Answer
Session Established
BYE
200 OK
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27
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 3262)
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28
Early Media
Phone
Unified CM
INVITE (no SDP)
183 Session Progress with SDP - Offer
PRACK with SDP - Answer
Media Stream Established
200 OK (PRACK)
200 OK (INVITE)
ACK
Session Established
BYE
200 OK
#CiscoLive
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29
Early Media w/ Early Offer
Phone
Unified CM
INVITE with SDP - Offer
183 Session Progress with SDP - Answer
PRACK (optional)
Media Stream Established
200 OK (PRACK) (optional)
200 OK (INVITE) with SDP (same as Answer in 183)
ACK
Session Established
BYE
200 OK
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30
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.
•
UPDATE method can also be used to re-negotiate prior to a final response.
#CiscoLive
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31
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 2018 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-CUCM12.0
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
Content-Length: 489
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32
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
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33
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 2018 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-CUCM12.0
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
#CiscoLive
DGTL-BRKUCC-2932
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34
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 2018 03:08:52 GMT
CSeq: 106 INVITE
Server: Cisco-8865/12.0.1
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-ciscoservice-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
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35
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
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36
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 2018 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
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37
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
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38
DTMF Relay
•
3 Methods for passing DTMF digits over a SIP network:
•
RFC 4733 (a.k.a. RFC 2833)
•
SIP NOTIFY
SIP Keypad Markup Language (KPML)
•
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39
DTMF Relay
RFC 4733 / RFC 2833
• Digits are passed in the RTP stream with a unique payload type
• Capability is negotiated in SDP like any other codec
Offer
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
#CiscoLive
Answer
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
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40
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 2017 14:48:00 GMT
Call-ID: 1a189580-1901fd20-962c99-3b6a12ac@172.18.106.59
... snip ...
Allow-Events: telephone-event
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
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41
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
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42
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 2017 14:48:11 GMT
User-Agent: Cisco-CUCM12.0
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
.d
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43
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 2017 15:05:24 GMT
Call-ID: 885e5780-19110134-96567f-3b6a12ac@172.18.106.59
User-Agent: Cisco-CUCM12.0
... snip ...
Allow-Events: presence, kpml
... snip ...
Session-Expires: 18000
Max-Forwards: 69
Content-Length: 0
#CiscoLive
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44
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 2017 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-15.6.2.T
Require: timer
Session-Expires: 18000;refresher=uac
Content-Type: multipart/mixed;boundary=uniqueBoundary
Mime-Version: 1.0
Content-Length: 600
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45
DTMF Relay
SIP KPML
Subscribe to 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-15.6.2.T
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:kpmlrequest kpml-request.xsd" version="1.0"><pattern persist="persist"><regex
tag="dtmf">[x*#ABCD]</regex></pattern></kpml-request>
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46
DTMF Relay
SIP KPML
Send a Digit
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-CUCM12.0
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"/>
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47
SIP Session-ID (RFC 7989)
•
Session-ID Header carries an end-to-end session identifier
•
Allows you to track a call as it traverses through various SIP systems
Session-ID: 65596d4800105000a00074a02fc0d796;remote=747a0ead00105000a00074a02fc0cf3b
Remote UUID
Local UUID
•
Session Identifier represented as {A,B} referring to {local,remote} UUIDs
•
{A,B} is equivalent to {B,A}
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48
SIP Call with Session-ID Headers
Phone 1
Unified CM
INVITE {A,N}
100 Trying {N,A}
200 OK {B,A}
ACK {A,B}
Phone 2
Phone 3
INVITE {A,N}
100 Trying {B,A}
200 OK {B,A}
ACK {A,B}
Session Established
REFER {B,A}
200 OK {A,B}
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49
SIP Call with Session-ID Headers
Phone 1
Unified CM
Phone 2
Phone 3
INVITE {A,N}
200 OK {C,A}
re-INVITE {C,A}
200 OK {A,C}
ACK {A,C}
Session Established
BYE {A,B}
200 OK {B,A}
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50
SIP Session-ID Support
Current Support
•
•
•
•
•
Unified CM 11.0 and later
Jabber 11.5 and later
78XX/88XX Endpoints 11.0 and later
CUBE 15.6(2)T and later
CVP 11.5 and later
•
Expressway X8.10 and later
CE and RoomOS Endpoints 9.6 and later
•
Webex Teams Hybrid Call Services
•
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51
Troubleshooting Tools
SIP Troubleshooting Tools
•
•
•
Unified CM / SME Tools:
•
Real Time Monitoring Tool / Session Trace
•
TranslatorX
•
Collaboration Solutions Analyzer
IOS (CUBE) and VCS Troubleshooting Tool:
•
TranslatorX
•
Collaboration Solutions Analyzer
Wireshark
#CiscoLive
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53
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 (uses calllogs file)
•
Session trace only traces SIP sessions (no SCCP, H.323, or MGCP)
•
Can display raw SIP messages
•
Uses correlation tags to include all call legs related to the call selected
#CiscoLive
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54
RTMT Session Trace Tool
#CiscoLive
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55
RTMT Session Trace Tool
Call Flow Diagram
#CiscoLive
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56
RTMT Session Trace Tool
Click on the message in the call flow diagram to see the actual message
#CiscoLive
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57
TranslatorX Tool
Features
•
Parses through Unified CM Trace Files (and CUBE, CUSP, VCS/Expressway, Jabber
10.x+, 88XX Phone Logs, PCAP Files (basic support))
•
Drag-and-Drop support for .txt as well as .gz files.
•
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 (including Session-ID Support)
•
Download for Windows, Mac OS X, and Linux from: http://translatorx.org/
•
NOTE: Do not call TAC for support on TranslatorX (although many TAC engineers
use it so feel free to mention you’re using it)
#CiscoLive
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58
TranslatorX Tool
#CiscoLive
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59
TranslatorX Tool – Call List Window
#CiscoLive
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60
TranslatorX Tool – Call List Filtering
Double-click for
complete Call Detail
Record
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61
TranslatorX Tool – CDR View
#CiscoLive
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62
TranslatorX Tool – CDR View
#CiscoLive
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63
TranslatorX Tool – Generating Filters
Select a Call and click
“Generate Filter” button
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64
TranslatorX Tool - Filters
#CiscoLive
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65
TranslatorX Tool – Call Flow Diagram
#CiscoLive
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66
TranslatorX Tool – Call Flow Diagram
#CiscoLive
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67
Collaboration Solutions Analyzer
cway.cisco.com/csa
#CiscoLive
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68
Wireshark
•
Open Source network packet capture and analysis tool
•
Available at http://www.wireshark.org
•
Available for Windows, Mac OS X, and UNIX/Linux
•
Provides VoIP Call and SIP analysis
#CiscoLive
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69
Wireshark
#CiscoLive
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70
Wireshark
VoIP Call Analysis
#CiscoLive
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71
Wireshark
VoIP Call Ladder Diagram
#CiscoLive
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72
Unified CM Trace
Configuration
Unified CM Trace Configuration
•
SIP messaging in Unified CM is written to the SDL trace file when
appropriate trace levels are set (SDI trace in for pre-9.0)
•
Configured from Cisco Unified Serviceability > Trace >
Configuration or by using Analysis Manager
•
Unified CM 9.0 and later combines SDI and SDL traces into the SDL
traces
•
Unified CM 9.0 and later fresh install and later default to detailed
tracing – no need to configure traces.
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74
Unified CM Trace Configuration
Select the Server
Select Service Group
Select the Service on
Which Trace Needs to
Be Enabled
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75
Unified CM Trace Configuration
1. Press Set Default
Updates All Servers
in this cluster with
these settings
2. Set to Detailed
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76
Unified CM Trace Configuration
#CiscoLive
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77
Trace Collection
Various Ways to Collect Trace Files
•
RTMT Collect Files
•
RTMT Analysis Manager
•
RTMT Remote Browse
•
RTMT Query Wizard
•
OS CLI (file get or file tail)
•
Recommended
file tail activelog cm/trace/ccm/sdl recent
#CiscoLive
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78
Gathering a Packet Capture from UCM
Use the Platform CLI command ‘utils network capture’
Reference
admin:utils network capture ? (or utils network capture-rotate)
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
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79
Expressway / VCS
Trace Configuration
VCS / Expressway Trace Configuration
•
VCS / Expressway-C / Expressway-E can log SIP messages to a
diagnostic log file.
•
To enable, navigate to Maintenance >
Diagnostics > Diagnostic logging
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81
VCS / Expressway Trace Configuration
Select this if you want to
get a Wireshark capture
Click Start new log
Click OK to Confirm
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82
VCS / Expressway Trace Configuration
Click Stop logging after
reproducing your problem
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83
VCS / Expressway Trace Configuration
Click to Collect the log and
tcpdump (for Wireshark)
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84
VCS / Expressway Trace Configuration
Click to Download the log and
tcpdump (for Wireshark)
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85
Cisco Unified Border
Element (CUBE) Trace
Configuration
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 debugging
no logging console
clock timezone EST -5 0
clock summer-time EDT recurring
ntp server 10.14.1.1
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87
CUBE Debugging
Various SIP debugs available
CUBE#debug ccsip ?
all
Enable
calls
Enable
dhcp
Enable
error
Enable
events
Enable
function
Enable
info
Enable
media
Enable
messages
Enable
non-call
Enable
preauth
Enable
states
Enable
translate Enable
transport Enable
verbose
Enable
all SIP debugging traces
CCSIP SPI calls debugging trace
SIP-DHCP debugging trace
SIP error debugging trace
SIP events debugging trace
SIP function debugging trace
SIP info debugging trace
SIP media debugging trace
CCSIP SPI messages debugging trace
Non-Call-Context trace (OPTIONS, SUBSCRIBE etc)
SIP preauth debugging traces
CCSIP SPI states debugging trace
SIP translation debugging trace
SIP transport debugging traces
verbose mode
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88
CUBE Logging Persistent
•
Continuously log debugs to flash / hard disk on CUBE
CUBE(config)#logging buffer 10000000 debugging
CUBE(config)#logging persistent url flash0:/MYDEBUGS size
104857600 filesize 5242880
Total Space Used
Size of Each File
Directory for Logs
Router#dir flash0:MYDEBUGS
Directory of flash0:/MYDEBUGS/
223
224
225
-rw-rw-rw-
5239168
5239631
3500757
Sep 27 2013 11:31:16 -04:00
Sep 27 2013 11:38:06 -04:00
Sep 27 2013 11:42:40 -04:00
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log_20170927-112427
log_20170927-113117
log_20170927-113807
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89
Cisco Unified Border Element Dial Peers
•
Incoming VoIP setup message from originating endpoint
•
This matches inbound VoIP dial peer 1 for characteristics such as codec, VAD,
DTMF method, protocol, etc.
•
Match the called number to outbound VoIP dial peer 2
•
Outgoing VoIP setup message
Originating
Endpoint
voice service voip
allow-connections sip to sip
Incoming VoIP Call
Outgoing VoIP Call
Terminating
Endpoint
CUBE
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
voice-class sip early-offer forced
dtmf-relay rtp-nte
codec g711ulaw
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90
CUBE show Commands
•
show call active voice [brief] shows state of currently active calls
0
: 13185070 3497557126ms.1 (19:31:29.719 EDT Fri Jun 8 2018) +2440 pid:200 Answer 9195555644 connected
dur 00:00:11 tx:535/107000 rx:466/93200 dscp:0 media:0 audio tos:0xB8 video tos:0x0
IP 173.37.124.198:57542 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off Transcoded: No ICE: 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
LostPacketRate:0.00 OutOfOrderRate:0.00
LocalUUID:c3ef30eaf83e77070c95a8a250e52fcd
RemoteUUID:ee064c2b6a275ea7a41bd022ab4f659c
VRF:
0
: 13185071 3497557126ms.2 (19:31:29.719 EDT Fri Jun 8 2018) +2440 pid:100 Originate 8008001180 connected
dur 00:00:11 tx:466/93200 rx:548/109600 dscp:0 media:0 audio tos:0xB8 video tos:0x0
IP 208.70.21.21:16016 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off Transcoded: No ICE: 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
LostPacketRate:0.00 OutOfOrderRate:0.00
LocalUUID:ee064c2b6a275ea7a41bd022ab4f659c
RemoteUUID:c3ef30eaf83e77070c95a8a250e52fcd
VRF:
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91
CUBE show Commands
•
show cube calls shows more specific CUBE-related call information
and allows filtering
CUBE# show cube calls all ?
callID
Display information matches callID
called-number
Display information matches called number
calling-number Display information matches calling number
conf-id
Display information matches conference ID
detail
Display detail level information
fpi-cor
Display information matches FPI Correlator
peer-callID
Display information matches peer callID
peer-rtp-port
Display information matches peer rtp-port number
rtp-port
Display information matches rtp-port number
|
Output modifiers
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92
CUBE show Commands
CUBE#show cube calls all called-number 8008001180
called number: 8008001180 info are as the following:
=============================================================
=============================================================
Phone number 8008001180 has the following callID associated to it:
=============================================================
CallID: 1730781, calling number: 9195551234
============================================================
A total of 1 call legs associated to number: 8008001180
============================================================
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93
CUBE show Commands
CUBE#show cube calls all callID 1730781
callid: 1730781 info are as the following:
=============================================================
SIP call leg info:
=============================================================
SIP CALL INFO of CCAPI callid 1730781
Call 1
SIP Call ID
: 41669A03-D5611E5-8A6384B0-5DED4CDB@64.102.250.104
State of the call
: STATE_ACTIVE (7)
Substate of the call
: SUBSTATE_NONE (0)
Calling Number
: 9195551234
Called Number
: 8008001180
Called URI
: sip:8008001180@208.70.21.21:5060
Bit Flags
: 0xE04018 0x90000100 0x0
CC Call ID
: 1730781
Source IP Address (Sig ): 64.102.250.104
Destn SIP Req Addr:Port : [208.70.21.21]:5060
Destn SIP Resp Addr:Port: [208.70.21.21]:5060
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94
CUBE – IOS-XE Embedded Traffic Capture
Reference
Export Packet Data in PCAP Format
• IP Traffic Export feature allows export of packets on an interface
• Configuration (From Enable mode, not Config mode):
monitor
monitor
monitor
monitor
capture
capture
capture
capture
capture-name
capture-name
capture-name
capture-name
access-list access-list-name
limit duration seconds
interface interface-name both
buffer circular size bytes
• Usage:
monitor capture capture-name start
monitor capture capture-name export file-location/file-name
monitor capture capture-name stop
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95
Reference
CUBE – IOS IP Traffic Capture
Export Packet Data in PCAP Format
• IP Traffic Export feature allows export of packets on an interface
• Configuration:
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
• Usage:
traffic-export interface g0/0 start
traffic-export interface g0/0 stop
traffic-export interface g0/0 copy scp://10.1.1.1/capture.pcap
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96
Case Studies
Case Study 1: Unable to Place a Call
Problem Description
•
A user reports that every time they call 80010001, they get reorder
(fast busy) tone
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98
Case Study 1: Unable to Place a Call
Use RTMT Session Trace
• Enter *80010001 into Called Number/URI field
• Set time and duration appropriately
• Search Finds one call
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99
Case Study 1: Unable to Place a Call
Use RTMT Session Trace
• Trace shows
signaling from both
phone and to
destination SIP trunk
• Receiving a 503
Service Unavailable
from destination
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100
Case Study 1: Unable to Place a Call
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.18.106.58:5060;branch=z9hG4bK34a0915e2c7a20
From: "Paul Giralt" <sip:89915644@172.18.106.58>;tag=5964355~0d0d25d7-4931-4a0783c6-b82e2c213ca7-44286097
To: <sip:80010001@10.81.98.203>;tag=932088316
Date: Sun, 07 Jun 2015 16:02:18 GMT
Call-ID: 9126a680-57416b0a-349ed0-3a6a12ac@172.18.106.58
CSeq: 101 INVITE
Allow-Events: presence
Warning: 399 collab-ccie-cm1a "Unable to find a device handler for the request received on
port 5060 from 172.18.106.58"
Content-Length: 0
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101
Case Study 2: 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
•
Called number is 1 877 288-8362
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102
Case Study 2: No One Answers the Phone
Collect Traces
• Problem is reproducible, so generate a test call and then collect traces.
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103
Case Study 2: No One Answers the Phone
Use TranslatorX to Analyze Traces
• Problem is reproducible, so generate a test call and then collect traces.
Drag and Drop folder into TranslatorX
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104
Case Study 2: No One Answers the Phone
Use TranslatorX to Analyze Traces
• Try to find call in Call List
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105
Case Study 2: No One Answers the Phone
Use TranslatorX to Analyze Traces
• Search for called party number
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106
Case Study 2: No One Answers the Phone
Use TranslatorX to Analyze Traces
• Disable Filters
• Select the INVITE
• Filter by SIP Call ID (control/command – S)
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107
Case Study 2: No One Answers the Phone
Use TranslatorX to Analyze Traces
• Where did the call originate? Try searching for the calling party number
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108
Case Study 2: 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)
• Re-enable Filters
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109
Case Study 2: No One Answers the Phone
Use TranslatorX to Analyze Traces
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110
Case Study 2: 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 2018 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,Xcisco-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
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Case Study 2: 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
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112
Case Study 2: 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 2018 14:36:33 GMT
Call-ID: 00260bd9-669e000b-588c0c2b-2193e2a3@172.18.159.152
CSeq: 101 INVITE
Allow-Events: presence
Content-Length: 0
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113
Case Study 2: 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-CUCM10.5
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
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114
Case Study 2: 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>
</x-cisco-remotecc-request>
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115
Case Study 2: 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 2018 14:36:33 GMT
User-Agent: Cisco-CUCM10.5
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>
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116
Case Study 2: 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:
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 2018 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:
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 2018 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
Content-Length: 0
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117
Case Study 2: No One Answers the Phone
Unified CM
IP Phone
(172.18.106.59)
(172.18.159.152)
CUBE
(172.18.159.231)
INVITE
100 Trying
REFER
SUBSCRIBE
200 OK (REFER)
200 OK (SUBSCRIBE)
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118
Case Study 2: 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 2018 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" digits="1" tag="Backspace OK"/>
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119
Case Study 2: 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 2018 14:36:34 GMT
Call-ID: 7747f400-bb01baf1-14686-3b6a12ac@172.18.106.59
CSeq: 1001 NOTIFY
Content-Length: 0
#CiscoLive
DGTL-BRKUCC-2932
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120
Case Study 2: 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
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-CUCM10.5
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
#CiscoLive
DGTL-BRKUCC-2932
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121
Case Study 2: 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>
</x-cisco-remotecc-request>
#CiscoLive
DGTL-BRKUCC-2932
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122
Case Study 2: 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 2018 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
#CiscoLive
DGTL-BRKUCC-2932
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123
Case Study 2: No One Answers the Phone
Unified CM
IP Phone
CUBE
(172.18.106.59)
(172.18.159.152)
(172.18.159.231)
INVITE
100 Trying
REFER
SUBSCRIBE
200 OK (REFER)
200 OK (SUBSCRIBE)
NOTIFY
200 OK (NOTIFY)
REFER
200 OK (REFER)
#CiscoLive
DGTL-BRKUCC-2932
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124
Case Study 2: No One Answers the Phone
User Dials a ‘8’
03/29/2010 10:36:34.944 |//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=z9hG4bK647d03c1
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 2018 14:36:34 GMT
CSeq: 1002 NOTIFY
Event: kpml
Subscription-State: active; expires=7195
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" digits="8" tag="Backspace OK"/>
#CiscoLive
DGTL-BRKUCC-2932
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125
Case Study 2: 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 2018 14:36:34 GMT
Call-ID: 7747f400-bb01baf1-14686-3b6a12ac@172.18.106.59
CSeq: 1001 NOTIFY
Content-Length: 0
#CiscoLive
DGTL-BRKUCC-2932
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126
Case Study 2: No One Answers the Phone
User Dials Remaining Digits
#CiscoLive
DGTL-BRKUCC-2932
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127
Case Study 2: No One Answers the Phone
Unified CM
IP Phone
CUBE
(172.18.106.59)
(172.18.159.152)
(172.18.159.231)
INVITE
100 Trying
REFER
SUBSCRIBE
200 OK (REFER)
200 OK (SUBSCRIBE)
NOTIFY
200 OK (NOTIFY)
REFER
200 OK (REFER)
NOTIFY
200 OK (NOTIFY)
NOTIFY / 200 OK
Repeats 10 Times
#CiscoLive
DGTL-BRKUCC-2932
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128
Case Study 2: No One Answers the Phone
CUCM Sends an INVITE to the CUBE
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 2018 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-CUCM10.5
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
Content-Length: 0
#CiscoLive
DGTL-BRKUCC-2932
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129
Case Study 2: No One Answers the Phone
Unified CM
IP Phone
CUBE
(172.18.106.59)
(172.18.159.152)
(172.18.159.231)
INVITE
100 Trying
REFER
SUBSCRIBE
200 OK (REFER)
200 OK (SUBSCRIBE)
NOTIFY
200 OK (NOTIFY)
REFER
200 OK (REFER)
NOTIFY
200 OK (NOTIFY)
NOTIFY / 200 OK
Repeats 10 Times
SUBSCRIBE
200 OK (SUBSCRIBE)
INVITE
#CiscoLive
DGTL-BRKUCC-2932
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130
Case Study 2: No One Answers the Phone
CUBE Replies With a 183 Session Progress W/ SDP
03/29/2010 10:36:42.324 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 1568 from
172.18.159.231:[5060]:
SIP/2.0 183 Session Progress
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>;tag=DE1EFF8-0
Date: Mon, 29 Mar 2018 14:37:23 GMT
Call-ID: 7c0ca800-bb01baf9-1468e-3b6a12ac@172.18.106.59
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:+18772888362@172.18.159.231>;party=called;screen=no;privacy=off
Contact: <sip:+18772888362@172.18.159.231:5060>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Length: 772
#CiscoLive
DGTL-BRKUCC-2932
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131
Case Study 2: No One Answers the Phone
CUBE Replies With a 183 Session Progress W/ SDP
v=0
o=CiscoSystemsSIP-GW-UserAgent 0 7954 IN IP4 172.18.159.231
s=SIP Call
c=IN IP4 172.18.159.231
t=0 0
m=audio 27980 RTP/AVP 0 8 116 18 100 101
c=IN IP4 172.18.159.231
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/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
#CiscoLive
DGTL-BRKUCC-2932
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132
Case Study 2: No One Answers the Phone
Unified CM Sends a 180 Ringing to the IP Phone
03/29/2010 10:36:42.330 |//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 172.18.159.152 on port
51682 index 2321
SIP/2.0 180 Ringing
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>;tag=97903bc0-a3de-4a15-ba27-44c81fe3adcd-45510542
Date: Mon, 29 Mar 2018 14:36:33 GMT
Call-ID: 00260bd9-669e000b-588c0c2b-2193e2a3@172.18.159.152
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence
Contact: <sip:9@172.18.106.59:5061;transport=tls>
Call-Info: <urn:x-cisco-remotecc:callinfo>; security= NotAuthenticated; orientation= to; ui-state= ringout; gci= 2305505; call-instance=1
Send-Info: conference
Remote-Party-ID: <sip:+18772888362@172.18.106.59>;party=called;screen=no;privacy=off
Content-Length: 0
#CiscoLive
DGTL-BRKUCC-2932
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133
Case Study 2: No One Answers the Phone
Unified CM
IP Phone
CUBE
(172.18.106.59)
(172.18.159.152)
(172.18.159.231)
INVITE
100 Trying
REFER
SUBSCRIBE
200 OK (REFER)
200 OK (SUBSCRIBE)
NOTIFY
200 OK (NOTIFY)
REFER
200 OK (REFER)
NOTIFY
200 OK (NOTIFY)
NOTIFY / 200 OK
Repeats 10 Times
SUBSCRIBE
INVITE
200 OK (SUBSCRIBE)
100 Trying
183 Session Progress
180 Ringing
#CiscoLive
DGTL-BRKUCC-2932
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134
Case Study 2: No One Answers the Phone
•
Phone Keeps Ringing
•
Timestamps Jump from 10:36:42 to 10:37:32
•
No SIP Signaling for 50 seconds
#CiscoLive
DGTL-BRKUCC-2932
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135
Case Study 2: No One Answers the Phone
Phone Sends a CANCEL
03/29/2010 10:37:32.934 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 172.18.159.152
on port 51682 index 2321 with 422 bytes:
CANCEL 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 2018 14:37:32 GMT
CSeq: 101 CANCEL
User-Agent: Cisco-CP9951/9.0.1
Content-Length: 0
#CiscoLive
DGTL-BRKUCC-2932
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136
Case Study 2: No One Answers the Phone
Unified CM Sends a 200 OK for the CANCEL
03/29/2010 10:37:32.935 |//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=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 2018 14:37:32 GMT
Call-ID: 00260bd9-669e000b-588c0c2b-2193e2a3@172.18.159.152
CSeq: 101 CANCEL
Content-Length: 0
#CiscoLive
DGTL-BRKUCC-2932
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137
Case Study 2: No One Answers the Phone
Unified CM
IP Phone
CUBE
(172.18.106.59)
(172.18.159.152)
(172.18.159.231)
NOTIFY
200 OK (NOTIFY)
CANCEL
200 OK (CANCEL)
CANCEL
487 Request Cancelled
200 OK (CANCEL)
487 Request Cancelled
ACK
ACK
#CiscoLive
DGTL-BRKUCC-2932
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138
Case Study 2: No One Answers the Phone
Unified CM
Phone 1
INVITE (w/ OFFER)
180 Ringing (no SDP)
SP SBC
SBC (CUBE)
SBC
INVITE (no SDP)
INVITE w/ OFFER
183 Session Progress (w/ OFFER)
#CiscoLive
SIP
SP
183 Session Progress (w/ ANSWER)
DGTL-BRKUCC-2932
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139
Case Study 2: No One Answers the Phone
Unified CM
Phone 1
INVITE (w/ OFFER)
SBC
INVITE (no SDP)
SIP
SP
INVITE w/ OFFER
183 Session Progress (w/ OFFER)
??? (w/ ANSWER)
SP SBC
SBC (CUBE)
183 Session Progress (w/ ANSWER)
??? (w/ ANSWER)
#CiscoLive
DGTL-BRKUCC-2932
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140
Case Study 2: Solution #1
Unified CM
Phone 1
INVITE (w/ OFFER)
SBC
INVITE (no SDP)
SIP
SP
INVITE w/ OFFER
183 Session Progress (w/ OFFER)
PRACK (w/ ANSWER)
SP SBC
SBC (CUBE)
183 Session Progress (w/ ANSWER)
PRACK (w/ ANSWER)
#CiscoLive
DGTL-BRKUCC-2932
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141
Case Study 2: Solution #2
Unified CM
Phone 1
INVITE (w/ OFFER)
SP SBC
SBC (CUBE)
SBC
INVITE (w/ OFFER)
SIP
SP
INVITE (w/ OFFER)
183 Session Progress (w/ ANSWER)
183 Session Progress (w/ ANSWER)
183 Session Progress (w/ ANSWER)
#CiscoLive
DGTL-BRKUCC-2932
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142
Case Study 2: No One Answers the Phone
Unified CM SIP Profile Configuration
#CiscoLive
DGTL-BRKUCC-2932
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143
Case Study 2: No One Answers the Phone
Unified CM
IP Phone
CUBE
(172.18.106.59)
(172.18.159.152)
(172.18.159.231)
INVITE
100 Trying
REFER
SUBSCRIBE
200 OK (REFER)
200 OK (SUBSCRIBE)
NOTIFY
200 OK (NOTIFY)
REFER
200 OK (REFER)
NOTIFY
200 OK (NOTIFY)
NOTIFY / 200 OK
Repeats 10 Times
SUBSCRIBE
INVITE
200 OK (SUBSCRIBE)
100 Trying
183 Session Progress
183 Session Progress
PRACK
#CiscoLive
DGTL-BRKUCC-2932
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144
Case Study 3: Calls to S4B Clients Fail
•
When a user dials a S4B client from a video-enabled Cisco phone,
the call fails. Call is from 58574 to 60051.
#CiscoLive
DGTL-BRKUCC-2932
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145
Case Study 3: Live
Demo
Case Study 3: Calls to S4B Clients Fail
M = {}
function M.outbound_INVITE(msg)
local contactHeader = msg:getHeader("Contact”)
if contactHeader then
local newContactHeader = string.gsub(contactHeader, ";video;audio;video", "")
msg:modifyHeader("Contact", newContactHeader)
end
end
return M
#CiscoLive
DGTL-BRKUCC-2932
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147
Case Study 3: Calls to S4B Clients Fail
•
For more information:
• Visit http://developer.cisco.com/web/sip/documentation to
download the SIP Normalization and Transparency Developer
Guide
#CiscoLive
DGTL-BRKUCC-2932
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148
Case Study 4: Intermittent Call Drops
•
Problem: Cisco TAC engineers are reporting to IT that they are
intermittently seeing some calls with customers dropping.
•
One call that was dropped came from +19715551000
#CiscoLive
DGTL-BRKUCC-2932
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149
Case Study 4: Live
Demo
Thank you
#CiscoLive
#CiscoLive