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Troubleshooting SIP with Cisco Unified Communications BRKUCC-2932
Paul Giralt
Distinguished Services Engineer
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Agenda
Introduction
Session Initiation Protocol (SIP) Overview
Troubleshooting Tools
Unified CM Tracing
Cisco Unified Border Element (CUBE) Tracing
Sample Call Flows / Case Studies
3
SIP Protocol Overview
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
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
Endpoints are referred to as User Agents
5
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
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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© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
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)
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© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
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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© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
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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© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
SIP INVITE Method
INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK1515b3154665 From: "Test User 1" <sip:[email protected]>;tag=97903bc0-43adcd-45510543 To: <sip:[email protected]> Call-ID: [email protected] 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:[email protected]> Contact: <sip:[email protected]:5060>;video;audio Max-Forwards: 69 Content-Length: 864 Content-Type: application/sdp
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© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
SIP Request Line
INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK1515b3154665 From: "Test User 1" <sip:[email protected]>;tag=97903bc0-43adcd-45510543 To: <sip:[email protected]> Call-ID: [email protected] 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:[email protected]> Contact: <sip:[email protected]:5060>;video;audio Max-Forwards: 69 Content-Length: 864 Content-Type: application/sdp
SIP Method
URI SIP Version
11
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
SIP Headers
INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK1515b3154665 From: "Test User 1" <sip:[email protected]>;tag=97903bc0-43adcd-45510543 To: <sip:[email protected]> Call-ID: [email protected] 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:[email protected]> Contact: <sip:[email protected]:5060>;video;audio Max-Forwards: 69 Content-Length: 864 Content-Type: application/sdp
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© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
SIP Response
SIP/2.0 404 Not Found Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bKb5291d44b969a4 From: "Paul Giralt" <sip:[email protected]>;tag=19210123ca7-45568313 To: <sip:[email protected]>;tag=253488-726 Date: Mon, 16 Jan 2012 04:00:22 GMT Call-ID: [email protected] CSeq: 101 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-15.2.2.T Reason: Q.850;cause=1 Content-Length: 0
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© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
SIP Response
SIP/2.0 404 Not Found Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bKb5291d44b969a4 From: "Paul Giralt" <sip:[email protected]>;tag=19210123ca7-45568313 To: <sip:[email protected]>;tag=253488-726 Date: Mon, 16 Jan 2012 04:00:22 GMT Call-ID: [email protected] 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
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SIP Response
SIP/2.0 404 Not Found Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bKb5291d44b969a4 From: "Paul Giralt" <sip:[email protected]>;tag=19210123ca7-45568313 To: <sip:[email protected]>;tag=253488-726 Date: Mon, 16 Jan 2012 04:00:22 GMT Call-ID: [email protected] CSeq: 101 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-15.2.2.T Reason: Q.850;cause=1 Content-Length: 0
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© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
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
603 Decline
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© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
INVITE
200 OK
Session Established
Phone 1 Unified CM
Basic SIP Call Setup
ACK
BYE
200 OK
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© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
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
CUBE
SBC (CUBE)
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© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
INVITE
200 OK
Phone 1
Basic SIP Call Setup with Unified CM and CUBE
ACK
BYE
200 OK
Unified CM
CUBE
SBC (CUBE)
INVITE
200 OK
ACK
BYE
200 OK
INVITE
200 OK
ACK
BYE
200 OK
SIP
SP SBC
SP SBC
Session Established
19
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
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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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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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 22
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
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
23
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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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INVITE with SDP - Offer
200 OK with SDP - Answer
Session Established
Phone 1 Unified CM
Early Offer
25
ACK (no SDP)
BYE
200 OK
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INVITE (no SDP)
200 OK with SDP - Offer
Session Established
Phone 1 Unified CM
Delayed Offer
26
ACK with SDP - Answer
BYE
200 OK
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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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INVITE (no SDP)
200 OK (INVITE) w/ SDP (should be same as answer)
Session Established
Phone 1 Unified CM
Early Media
28
ACK
BYE
200 OK
183 Session Progress with SDP - Offer
PRACK with SDP - Answer
Media Stream Established
200 OK (PRACK)
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Media Re-negotiation Re-INVITE
29
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.
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Media Re-negotiation Re-INVITE
30
INVITE sip:[email protected]:49833;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK901f9c72c19221 From: "Paul Giralt" <sip:[email protected]>;tag=15462272~0d0d25d7-4931-4a07-83c6-b82e2c213ca7-45545776 To: <sip:[email protected]>;tag=0022bdd6843100702aae8e5b-4be253be Date: Wed, 11 Jan 2012 03:08:51 GMT Call-ID: [email protected] 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:[email protected]>;party=calling;screen=yes;privacy=off Contact: <sip:[email protected]:5061;transport=tls> Content-Type: application/sdp Content-Length: 489
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Media Re-negotiation Re-INVITE – Stopping a Media Session
31
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
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Media Re-negotiation Re-INVITE – Delayed Offer to Re-establish Media Stream
32
INVITE sip:[email protected]:49833;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK901fac34c0fb1b From: "Paul Giralt" <sip:[email protected]>;tag=15462272~0d0d25d7-4931-4a07-83c6-b82e2c213ca7-45545776 To: <sip:[email protected]>;tag=0022bdd6843100702aae8e5b-4be253be Date: Wed, 11 Jan 2012 03:08:52 GMT Call-ID: [email protected] 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:[email protected]>;party=calling;screen=yes;privacy=off Contact: <sip:[email protected]:5061;transport=tls> Content-Length: 0
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Media Re-negotiation Re-INVITE – Offer in 200 OK
33
SIP/2.0 200 OK Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK901fac34c0fb1b From: "Paul Giralt" <sip:[email protected]>;tag=15462272~0d0d25d7-4931-4a07-83c6-b82e2c213ca7-45545776 To: <sip:[email protected]>;tag=0022bdd6843100702aae8e5b-4be253be Call-ID: [email protected] Date: Wed, 11 Jan 2012 03:08:52 GMT CSeq: 106 INVITE Server: Cisco-CPCIUS/9.2.1 Contact: <sip:[email protected]:49833;transport=tls> Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO Remote-Party-ID: "Paul Giralt" <sip:[email protected]>;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
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Media Re-negotiation Re-INVITE – Offer in 200 OK
34
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
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Media Re-negotiation Re-INVITE – Answer in ACK
35
ACK sip:[email protected]:49833;transport=tls SIP/2.0 Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK901fb064465a06 From: "Paul Giralt" <sip:[email protected]>;tag=15462272~0d0d25d7-4931-4a07-83c6-b82e2c213ca7-45545776 To: <sip:[email protected]>;tag=0022bdd6843100702aae8e5b-4be253be Date: Wed, 11 Jan 2012 03:08:52 GMT Call-ID: [email protected] Max-Forwards: 70 CSeq: 106 ACK Allow-Events: presence Content-Type: application/sdp Content-Length: 446
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Media Re-negotiation Re-INVITE – Answer in ACK – Decline Video Support
36
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
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
DTMF Relay
3 Methods for passing DTMF digits over a SIP network:
– RFC 2833
– SIP NOTIFY
– SIP Keypad Markup Language (KPML)
37
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DTMF Relay RFC 2833
38
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
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DTMF Relay SIP NOTIFY
39
Passes DTMF information in a SIP NOTIFY message telephone-event Event
Negotiated in Call-Info header
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK9843c455840434
From: "Paul Giralt" <sip:[email protected]>;tag=14902469~0d0d25d7-4931-4a07-83c6
To: <sip:[email protected]>
Date: Mon, 13 May 2013 14:48:00 GMT
Call-ID: [email protected]
... 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
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
DTMF Relay SIP NOTIFY
40
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK9843c455840434
From: "Paul Giralt" <sip:[email protected]>;tag=14902469~0d0d25d7-4931-4a07-83c6
To: <sip:[email protected]>;tag=4363A830-17FC
Call-ID: [email protected]
... 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
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DTMF Relay SIP NOTIFY
41
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:[email protected]>;tag=14902469~0d0d25d7-4931-4a07-83c6
To: <sip:[email protected]>;tag=4363A830-17FC
Call-ID: [email protected]
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:[email protected]>
Content-Type: audio/telephone-event
Content-Length: 4
.d
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
DTMF Relay SIP KPML
42
Passes DTMF information in a SIP NOTIFY message kpml Event
Capability advertised in Allow-Events – uses SUBSCRIBE message to subscribe
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK986efd6c4e51e4
From: "Paul Giralt" <sip:[email protected]>;tag=14918970~0d0d25d7-4931-4a07-83c6
To: <sip:[email protected]>
Date: Mon, 13 May 2013 15:05:24 GMT
Call-ID: [email protected]
User-Agent: Cisco-CUCM10.0
... snip ...
Allow-Events: presence, kpml
... snip ...
Session-Expires: 18000
Max-Forwards: 69
Content-Length: 0
Offer
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
DTMF Relay SIP KPML
43
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK986efd6c4e51e4
From: "Paul Giralt" <sip:[email protected]>;tag=14918970~0d0d25d7-4931-4a07-83c6
To: <sip:[email protected]>;tag=437394E8-2E1
Date: Mon, 13 May 2013 15:05:26 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO
Allow-Events: kpml, telephone-event
Remote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=off
Contact: <sip:[email protected]: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
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DTMF Relay SIP KPML
44
SUBSCRIBE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.106.231:5060;branch=z9hG4bKBAE27139E
From: <sip:[email protected]>;tag=437394E8-2E1
To: "Paul Giralt" <sip:[email protected]>;tag=14918970~0d0d25d7-4931-4a07-83c6
Call-ID: [email protected]
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-request.xsd" version="1.0"><pattern persist="persist"><regex
tag="dtmf">[x*#ABCD]</regex></pattern></kpml-request>
Subscribe to KPML
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DTMF Relay SIP KPML
45
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:[email protected]>;tag=14918970~0d0d25d7-4931-4a07-83c6
To: <sip:[email protected]>;tag=437394E8-2E1
Call-ID: [email protected]
CSeq: 104 NOTIFY
Max-Forwards: 70
User-Agent: Cisco-CUCM10.0
Event: kpml
Subscription-State: active;expires=7197
Contact: <sip:[email protected]: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"/>
Send a Digit
Troubleshooting Tools
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SIP Troubleshooting Tools
Unified CM / SME Tools:
– Real Time Monitoring Tool / Session Trace
– TranslatorX
Wireshark
47
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RTMT Session Trace Tool Session Trace Features
48
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.
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
RTMT Session Trace Tool
49
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RTMT Session Trace Tool
Call Flow Diagram
50
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RTMT Session Trace Tool
Click on the message in the call flow diagram to see the actual message
51
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TranslatorX Tool Features
52
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)
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
TranslatorX Tool
53
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TranslatorX Tool Call List Window
54
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TranslatorX Tool Call List Filtering
55
Double-click for complete
Call Detail Record
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TranslatorX Tool CDR View
56
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TranslatorX Tool CDR View
57
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
TranslatorX Tool Call List Filtering
58
Select a Call and click
“Generate Filter” button
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TranslatorX Tool Message Filters
59
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TranslatorX Tool
60
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
TranslatorX Tool
61
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
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
62
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Wireshark
63
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Wireshark VoIP Call Analysis
64
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Wireshark VoIP Call Ladder Diagram
65
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Wireshark How to Gather a Trace?
66
Both Unified CM and IOS provide a mechanism to gather a packet capture
Will be covered later
Unified CM Tracing Configuration
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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 traces.
68
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Unified CM Trace Configuration
69
Select the
Server
Select the Service on
Which Trace Needs to
Be Enabled
Select Service
Group
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Unified CM Trace Configuration
70
1. Press
Set Default
2. Set to Detailed
Updates All
Servers in This
Cluster with
These Settings
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Unified CM Trace Configuration
71
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
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Unified CM Trace Configuration
72
Can Also Use the Troubleshooting Trace Settings Page in CallManager Serviceability (Trace > Troubleshooting Trace Settings)
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Trace Collection Various Ways to Collect Trace Files
73
RTMT Analysis Manager
RTMT Remote Browse
RTMT Collect Files
RTMT Query Wizard
OS CLI (file get or file tail)
Recommended
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Gathering a Packet Capture from Unified CM
Use the Platform CLI command ‘utils network capture’
74
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
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CUBE Debugging
CUBE / IOS Tools:
– IOS debugs
– IOS show commands
– Per-call trace
– Packet export
76
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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
77
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
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CUBE Debugging
Various SIP debugs available:
78
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
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CUBE Debugging Sample ‘debug ccsip messages’
79
Jan 12 03:14:43.102: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 172.18.106.59:5060;branch=z9hG4bK978d2e8df73dc
From: "Paul Giralt" <sip:[email protected]>;tag=16218435~-b82e2c213ca7-45552048
To: <sip:[email protected]>
Date: Thu, 12 Jan 2012 03:09:42 GMT
Call-ID: [email protected]
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:[email protected]>
Remote-Party-ID: "Paul Giralt" <sip:[email protected]>;party=calling;screen=yes;privacy=off
Contact: <sip:[email protected]:5060;transport=tcp>;video;audio
Max-Forwards: 69
Content-Length: 0
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
CUBE Debugging
Other generic voice debugs can be useful as well:
– debug voice ccapi inout
– debug voice dialpeer
– debug voice rtp session dtmf-relay
– debug voice rtp session named-event (for any RFC 2833 packets)
80
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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
81
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
CUBE
voice service voip
allow-connections sip to sip
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
CUBE show Commands
show call active voice [brief] shows state of currently active calls
82
0 : 2807 92135710ms.1 (23:55:20.115 EST Mon Jan 16 2012) +1770 pid:1 Answer 89915644 active
dur 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
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
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
– Call Admission Control limits
83
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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
84
Reference
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CUBE – IP Traffic Capture Export Packet Data in PCAP Format
85
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
traffic-export interface g0/0 copy scp://10.1.1.1/capture.pcap
Case Studies
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Case Study 1: Unable to place a call Problem Description
87
A user reports that every time they call (919) 555-1212, they get a message that the call could not be completed as dialed.
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 1: Unable to Place a Call Use RTMT Session Trace
88
Enter *5551212 into Called Number/URI field
Set time and duration appropriately
Search Finds two calls
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Case Study 1: Unable to Place a Call Use RTMT Session Trace
89
Double-click to see message diagram
Clearly shows the far-end sends back a 404 Not Found
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Case Study 1: Unable to Place a Call Troubleshoot Call on CUBE
90
Enable SIP message debugs – debug ccsip messages Jan 16 04:00:22.679: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bKb5291d44b969a4
From: "Paul Giralt" <sip:[email protected]>;tag=19210128~0d0d25d7-4931-4a07-83c6-b82e2c213ca7-45568313
To: <sip:[email protected]>
Date: Mon, 16 Jan 2012 03:55:17 GMT
Call-ID: [email protected]
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:[email protected]>
Contact: <sip:[email protected]:5060>
Max-Forwards: 69
Content-Length: 0
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 1: Unable to Place a Call Troubleshoot Call on CUBE
91
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:[email protected]>;tag=19210128~0d0d25d7-4931-4a07-83c6-b82e2c213ca7-
45568313
To: <sip:[email protected]>;tag=253488-726
Date: Mon, 16 Jan 2012 04:00:22 GMT
Call-ID: [email protected]
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
No match, result=-1
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Case Study 2: Unable to Place a Call #2 Problem Description
92
A user reports that every time they call (919) 555-1212, they get reorder (fast busy) tone.
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 2: Unable to Place a Call #2 Use RTMT Session Trace
93
Enter *5551212 into Called Number/URI field
Set time and duration appropriately
Search Finds two calls
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Case Study 2: Unable to Place a Call #2 Use RTMT Session Trace
94
Trace shows signaling from phone as well as to CUBE
CUBE is responding with a 403 Forbidden
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Case Study 2: Unable to Place a Call #2 Problem Description
95
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
<- PREFERRED (More Secure)
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone Problem Description
96
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
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone Collect Traces
97
Problem is reproducible, so generate a test call and then collect traces.
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Case Study 3: No One Answers the Phone Use TranslatorX
98
Problem is reproducible, so generate a test call and then collect traces. Select File > Open Folder…
Or just drag and drop the folder of traces into the translator window.
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone Use TranslatorX to Analyze Traces
99
Search for called party number
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone Use TranslatorX to Analyze Traces
100
Disable Filters
Select the INVITE
Filter by SIP Call ID (control/command – S)
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone Use TranslatorX to Analyze Traces
101
03/29/2010 10:36:41.497 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 172.18.159.231:[5060]: INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK1515b3154665 From: "Test User 1" <sip:[email protected]>;tag=97903bc0-a3de-4a15-ba27-44c81fe3adcd-45510543 To: <sip:[email protected]> Date: Mon, 29 Mar 2010 14:36:41 GMT Call-ID: [email protected] 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:[email protected]> Contact: <sip:[email protected]:5060>;video;audio Max-Forwards: 69 Content-Length: 0
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone Use TranslatorX to Analyze Traces
102
Where did the call originate? Try searching for the calling party number
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone Use TranslatorX to Analyze Traces
103
Select the INVITE
Create New Filter (control/command-N)
Filter by IP Address (control/command – I)
Re-enable Filters
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Case Study 3: No One Answers the Phone Use TranslatorX to Analyze Traces
104
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Case Study 3: No One Answers the Phone INVITE from IP Phone w/ SDP
105
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:[email protected];user=phone SIP/2.0 Via: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK1636ab61 From: "Test User 1" <sip:[email protected]>;tag=00260bd9669e07147bcb3aac-3cda8f0c To: <sip:[email protected];user=phone> Call-ID: [email protected] Max-Forwards: 70 Date: Mon, 29 Mar 2010 14:36:33 GMT CSeq: 101 INVITE User-Agent: Cisco-CP9951/9.0.1 Contact: <sip:[email protected]: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:[email protected]>;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
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone INVITE from IP Phone w/ SDP (continued)
106
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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Case Study 3: No One Answers the Phone Unified CM Sends a 100 Trying
107
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:[email protected]>;tag=00260bd9669e07147bcb3aac-3cda8f0c To: <sip:[email protected];user=phone> Date: Mon, 29 Mar 2010 14:36:33 GMT Call-ID: [email protected] CSeq: 101 INVITE Allow-Events: presence Content-Length: 0
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Case Study 3: No One Answers the Phone Unified CM Sends a REFER to Play Outside Dialtone
108
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:[email protected]:51682 SIP/2.0 Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK151511c5f04bf From: <sip:[email protected]>;tag=2144536187 To: <sip:[email protected]> Call-ID: [email protected] CSeq: 101 REFER Max-Forwards: 70 Contact: <sip:[email protected]:5061;transport=tls> User-Agent: Cisco-CUCM8.0 Expires: 0 Refer-To: cid:[email protected] Content-Id: <[email protected]> Require: norefersub Content-Type: application/x-cisco-remotecc-request+xml Referred-By: <sip:[email protected]> Content-Length: 409
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Case Study 3: No One Answers the Phone Unified CM Sends a REFER to play Outside Dialtone (continued)
109
<x-cisco-remotecc-request> <playtonereq> <dialogid> <callid>[email protected]</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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Case Study 3: No One Answers the Phone Unified CM Sends a SUBSCRIBE for KPML
110
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:[email protected]:51682 SIP/2.0 Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK1515232b4e84f From: <sip:[email protected]>;tag=1976165806 To: <sip:[email protected]> Call-ID: [email protected] CSeq: 101 SUBSCRIBE Date: Mon, 29 Mar 2010 14:36:33 GMT User-Agent: Cisco-CUCM8.0 Event: kpml; [email protected]; from-tag=00260bd9669e07147bcb3aac-3cda8f0c Expires: 7200 Contact: <sip:[email protected]: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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Case Study 3: No One Answers the Phone Phone Sends 200 OK for the REFER and SUBSCRIBE
111
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:[email protected]>;tag=2144536187
To: <sip:[email protected]>;tag=00260bd9669e07167c743311-343ee3af
Call-ID: [email protected]
Date: Mon, 29 Mar 2010 14:36:33 GMT
CSeq: 101 REFER
Server: Cisco-CP9951/9.0.1
Contact: <sip:[email protected]: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:[email protected]>;tag=1976165806
To: <sip:[email protected]>;tag=00260bd9669e07177ee0d51d-14f56f89
Call-ID: [email protected]
Date: Mon, 29 Mar 2010 14:36:33 GMT
CSeq: 101 SUBSCRIBE
Server: Cisco-CP9951/9.0.1
Contact: <sip:[email protected]:51682;transport=TLS>
Expires: 7200
Content-Length: 0
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
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)
112
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Case Study 3: No One Answers the Phone User Dials a ‘1’
113
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:[email protected]:5061 SIP/2.0 Via: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK1cd529ba To: <sip:[email protected]>;tag=1976165806 From: <sip:[email protected]>;tag=00260bd9669e07177ee0d51d-14f56f89 Call-ID: [email protected] Date: Mon, 29 Mar 2010 14:36:33 GMT CSeq: 1001 NOTIFY Event: kpml Subscription-State: active; expires=7200 Max-Forwards: 70 Contact: <sip:[email protected]: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"/>
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone Unified CM Replies to NOTIFY With a 200 OK
114
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:[email protected]>;tag=00260bd9669e07177ee0d51d-14f56f89 To: <sip:[email protected]>;tag=1976165806 Date: Mon, 29 Mar 2010 14:36:34 GMT Call-ID: [email protected] CSeq: 1001 NOTIFY Content-Length: 0
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone Unified CM Replies Sends a REFER to Disable Outside Dialtone
115
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:[email protected]:51682 SIP/2.0 Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK151536ea86ab0 From: <sip:[email protected]>;tag=1574166193 To: <sip:[email protected]> Call-ID: [email protected] CSeq: 101 REFER Max-Forwards: 70 Contact: <sip:[email protected]:5061;transport=tls> User-Agent: Cisco-CUCM8.0 Expires: 0 Refer-To: cid:[email protected] Content-Id: <[email protected]> Require: norefersub Content-Type: application/x-cisco-remotecc-request+xml Referred-By: <sip:[email protected]> Content-Length: 401
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone
116
<x-cisco-remotecc-request> <playtonereq> <dialogid> <callid>[email protected]</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>
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone Phone Replies With 200 OK to REFER
117
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:[email protected]>;tag=1574166193 To: <sip:[email protected]>;tag=00260bd9669e07184b08b96b-796ab86f Call-ID: [email protected] Date: Mon, 29 Mar 2010 14:36:33 GMT CSeq: 101 REFER Server: Cisco-CP9951/9.0.1 Contact: <sip:[email protected]:51682;transport=TLS> Content-Length: 0
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone
118
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)
NOTIFY
200 OK (NOTIFY)
REFER
200 OK (REFER)
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone User Dials a ‘8’
119
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:[email protected]:5061 SIP/2.0 Via: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK647d03c1 To: <sip:[email protected]>;tag=1976165806 From: <sip:[email protected]>;tag=00260bd9669e07177ee0d51d-14f56f89 Call-ID: [email protected] Date: Mon, 29 Mar 2010 14:36:34 GMT CSeq: 1002 NOTIFY Event: kpml Subscription-State: active; expires=7195 Max-Forwards: 70 Contact: <sip:[email protected]: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"/>
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone Unified CM Replies to NOTIFY With a 200 OK
120
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:[email protected]>;tag=00260bd9669e07177ee0d51d-14f56f89 To: <sip:[email protected]>;tag=1976165806 Date: Mon, 29 Mar 2010 14:36:34 GMT Call-ID: [email protected] CSeq: 1001 NOTIFY Content-Length: 0
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone User Dials Remaining Digits
121
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
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)
NOTIFY
200 OK (NOTIFY)
REFER
200 OK (REFER)
NOTIFY
200 OK (NOTIFY)
NOTIFY / 200 OK
Repeats 10 Times
122
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone Digit Analysis Match
123
10:36:41.486 |Digit analysis: match(pi="2", fqcn="+19194769236", cn="89919236",plv="5", pss="1stLine:RTP_AbbrDial:Cisco:US Local:US RTP
Local:US Long Distance:US International:VMPilotPartition", TodFilteredPss="1stLine:RTP_AbbrDial:Cisco:US Local:US RTP Local:US Long
Distance:US International:VMPilotPartition", dd="918772888362",dac="1”)
10:36:41.486 |Digit analysis: analysis results
10:36:41.486 ||PretransformCallingPartyNumber=+19194769236
|CallingPartyNumber=+19194769236
|DialingPartition=GDP_GlobalE164_PSTN
|DialingPattern=\+1.[2-9]XX[2-9]XXXXXX
|FullyQualifiedCalledPartyNumber=+18772888362
|DialingPatternRegularExpression=(+1)([2-9][0-9][0-9][2-9][0-9][0-9][0-9][0-9][0-9][0-9])
|DialingWhere=
|PatternType=Enterprise
|PotentialMatches=NoPotentialMatchesExist
|DialingSdlProcessId=(0,0,0)
|PretransformDigitString=+18772888362
|PretransformTagsList=ACCESS-CODE:SUBSCRIBER
|PretransformPositionalMatchList=+1:8772888362
|CollectedDigits=+18772888362
|UnconsumedDigits=
|TagsList=ACCESS-CODE:SUBSCRIBER
|PositionalMatchList=+1:8772888362
|VoiceMailbox=
|VoiceMailCallingSearchSpace=1stLine:RTP_AbbrDial
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone Digit Analysis Match
124
|VoiceMailPilotNumber=89944444
|RouteBlockFlag=RouteThisPattern
|RouteBlockCause=0
|AlertingName=
|UnicodeDisplayName=
|DisplayNameLocale=1
|OverlapSendingFlagEnabled=0
|WithTags=
|WithValues=
|CallingPartyNumberPi=NotSelected
|ConnectedPartyNumberPi=NotSelected
|CallingPartyNamePi=NotSelected
|ConnectedPartyNamePi=NotSelected
|CallManagerDeviceType=NoDeviceType
|PatternPrecedenceLevel=Routine
|CallableEndPointName=[23146446-6606-7227-3882-75d07dd6fdef]
|PatternNodeId=[9badd465-d20a-5bc7-1077-8edee47e8caf]
|AARNeighborhood=[]
|AARDestinationMask=[]
|AARKeepCallHistory=true
|AARVoiceMailEnabled=false
|NetworkLocation=OffNet
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone Digit Analysis Match
125
|Calling Party Number Type=Cisco Unified CallManager
|Calling Party Numbering Plan=Cisco Unified CallManager
|Called Party Number Type=Cisco Unified CallManager
|Called Party Numbering Plan=Cisco Unified CallManager
|ProvideOutsideDialtone=false
|AllowDeviceOverride=false
|AlternateMatches=
{
|Partition=US Long Distance
{
<
|Pattern=9.1[2-9]XX[2-9]XXXXXX
|PatternType=Translation
|TranslationPartition=[a6bd708e-ac4d-ae55-3134-b90b987e5ad9]
|CallManagerDeviceType=NoDeviceType
|PatternPrecedenceLevel=PlDefault
|PatternRouteClass=RouteClassDefault
|RouteNextHopByCgpn=false
>
}
}
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone Digit Analysis Match
126
|TranslationPatternDetails=
|PretransformCallingPartyNumber=89919236
|CallingPartyNumber=+19194769236
|DialingPartition=US Local
|DialingPattern=9.1877[2-9]XXXXXX
|FullyQualifiedCalledPartyNumber=918772888362
|DialingPatternRegularExpression=(9)(1877[2-9][0-9][0-9][0-9][0-9][0-9][0-9])
|DialingWhere=
|PatternType=Translation
|PotentialMatches=NoPotentialMatchesExist
|DialingSdlProcessId=(0,0,0)
|PretransformDigitString=918772888362
|PretransformTagsList=ACCESS-CODE:SUBSCRIBER
|PretransformPositionalMatchList=9:18772888362
|CollectedDigits=+18772888362
|UnconsumedDigits=
|TagsList=SUBSCRIBER
|PositionalMatchList=18772888362
|VoiceMailbox=
|VoiceMailCallingSearchSpace=
|VoiceMailPilotNumber=
|RouteBlockFlag=RouteThisPattern
|RouteBlockCause=1
|AlertingName=
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone Digit Analysis Match
127
|UnicodeDisplayName=
|DisplayNameLocale=1
|OverlapSendingFlagEnabled=0
|WithTags=
|WithValues=
|CallingPartyNumberPi=NotSelected
|ConnectedPartyNumberPi=NotSelected
|CallingPartyNamePi=NotSelected
|ConnectedPartyNamePi=NotSelected
|CallManagerDeviceType=NoDeviceType
|PatternPrecedenceLevel=Routine
|CallableEndPointName=[bb6f140a-5fd4-179a-2cad-2a1d5eacca7e]
|PatternNodeId=[bb6f140a-5fd4-179a-2cad-2a1d5eacca7e]
|AARNeighborhood=[]
|AARDestinationMask=[]
|AARKeepCallHistory=true
|AARVoiceMailEnabled=false
|NetworkLocation=OnNet
|Calling Party Number Type=Cisco Unified CallManager
|Calling Party Numbering Plan=Cisco Unified CallManager
|Called Party Number Type=Cisco Unified CallManager
|Called Party Numbering Plan=Cisco Unified CallManager
|ProvideOutsideDialtone=true
|AllowDeviceOverride=false
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone CUCM Sends an INVITE to the CUBE
128
03/29/2010 10:36:41.497 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 172.18.159.231:[5060]: INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK1515b3154665 From: "Test User 1" <sip:[email protected]>;tag=97903bc0-a3de-4a15-ba27-44c81fe3adcd-45510543 To: <sip:[email protected]> Date: Mon, 29 Mar 2010 14:36:41 GMT Call-ID: [email protected] 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:[email protected]> Contact: <sip:[email protected]:5060>;video;audio Max-Forwards: 69 Content-Length: 0
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
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)
NOTIFY
200 OK (NOTIFY)
REFER
200 OK (REFER)
NOTIFY
200 OK (NOTIFY)
NOTIFY / 200 OK
Repeats 10 Times
SUBSCRIBE
INVITE 200 OK (SUBSCRIBE)
129
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone CUBE Replies With a 183 Session Progress W/ SDP
130
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:[email protected]>;tag=97903bc0-a3de-4a15-ba27-44c81fe3adcd-45510543 To: <sip:[email protected]>;tag=DE1EFF8-0 Date: Mon, 29 Mar 2010 14:37:23 GMT Call-ID: [email protected] CSeq: 101 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=off Contact: <sip:[email protected]:5060> Supported: sdp-anat Server: Cisco-SIPGateway/IOS-12.x Content-Type: multipart/mixed;boundary=uniqueBoundary Mime-Version: 1.0 Content-Length: 788 --uniqueBoundary
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone CUBE Replies With a 183 Session Progress W/ SDP
131
Content-Type: application/sdp Content-Disposition: session;handling=required 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 --uniqueBoundary Content-Type: application/x-q931 Content-Disposition: signal;handling=optional Content-Length: 11
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Case Study 3: No One Answers the Phone Unified CM Sends a 180 Ringing to the IP Phone
132
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:[email protected]>;tag=00260bd9669e07147bcb3aac-3cda8f0c To: <sip:[email protected];user=phone>;tag=97903bc0-a3de-4a15-ba27-44c81fe3adcd-45510542 Date: Mon, 29 Mar 2010 14:36:33 GMT Call-ID: [email protected] CSeq: 101 INVITE Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY Allow-Events: presence Contact: <sip:[email protected]:5061;transport=tls> Call-Info: <urn:x-cisco-remotecc:callinfo>; security= NotAuthenticated; orientation= to; ui-state= ringout; gci= 2-305505; call-instance= 1 Send-Info: conference Remote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=off Content-Length: 0
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
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)
NOTIFY
200 OK (NOTIFY)
REFER
200 OK (REFER)
NOTIFY
200 OK (NOTIFY)
NOTIFY / 200 OK
Repeats 10 Times
SUBSCRIBE
INVITE
100 Trying
200 OK (SUBSCRIBE)
180 Ringing
183 Session Progress
133
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Case Study 3: No One Answers the Phone Phone Keeps Ringing
134
Timestamps Jump from 10:36:42 to 10:37:32
No SIP Signaling for 50 seconds
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Case Study 3: No One Answers the Phone Phone Sends a CANCEL
135
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:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK1636ab61
From: "Test User 1" <sip:[email protected]>;tag=00260bd9669e07147bcb3aac-3cda8f0c
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
Max-Forwards: 70
Date: Mon, 29 Mar 2010 14:37:32 GMT
CSeq: 101 CANCEL
User-Agent: Cisco-CP9951/9.0.1
Content-Length: 0
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Case Study 3: No One Answers the Phone Unified CM Sends a 200 OK for the CANCEL
136
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:[email protected]>;tag=00260bd9669e07147bcb3aac-3cda8f0c
To: <sip:[email protected];user=phone>
Date: Mon, 29 Mar 2010 14:37:32 GMT
Call-ID: [email protected]
CSeq: 101 CANCEL
Content-Length: 0
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone
137
IP Phone (172.18.159.152)
Unified CM (172.18.159.152)
CUBE (172.18.159.231)
NOTIFY
200 OK (NOTIFY)
CANCEL
200 OK (CANCEL)
CANCEL
200 OK (CANCEL)
487 Request Cancelled
487 Request Cancelled
ACK
ACK
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183 Session Progress (w/ ANSWER)
Phone 1
Case Study 3: No One Answers the Phone
Unified CM
CUBE
SBC (CUBE) SIP
SP SBC
SP SBC
138
INVITE w/ OFFER INVITE (no SDP)
INVITE (w/ OFFER)
183 Session Progress (w/ OFFER) 180 Ringing (no SDP)
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
183 Session Progress (w/ ANSWER)
Phone 1
Case Study 3: No One Answers the Phone
Unified CM
CUBE
SBC (CUBE) SIP
SP SBC
SP SBC
139
INVITE w/ OFFER INVITE (no SDP)
INVITE (w/ OFFER)
183 Session Progress (w/ OFFER)
??? (w/ ANSWER) ??? (w/ ANSWER)
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183 Session Progress (w/ ANSWER)
Phone 1
Case Study 3: No One Answers the Phone
Unified CM
CUBE
SBC (CUBE) SIP
SP SBC
SP SBC
140
INVITE w/ OFFER INVITE (no SDP)
INVITE (w/ OFFER)
183 Session Progress (w/ OFFER)
183 Session Progress (w/ ANSWER) PRACK (w/ ANSWER)
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone
How do we get the gateway to cut through audio on the 183 Session Progress message?
RFC 3262: Reliability of Provisional Responses in the Session Initiation Protocol (SIP)
Provides a way to acknowledge the 183 Session Progress message – PRACK
Unified CM parameter “SIP Rel1XX Options” *
–Disabled
–Send PRACK for all 1xx Messages
–Send PRACK if 1xx Contains SDP
141
*Service Parameter in 7.x and earlier. SIP Profile parameter in 8.x and later
cube(conf-serv-sip)#rel1xx ?
disable Disables reliable-provisional responses
require Requires reliable-provisional responses
supported Supports reliable-provisional responses
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone Unified CM SIP Profile Configuration
142
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 3: No One Answers the Phone
143
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)
NOTIFY
200 OK (NOTIFY)
REFER
200 OK (REFER)
NOTIFY
200 OK (NOTIFY)
NOTIFY / 200 OK
Repeats 10 Times
SUBSCRIBE
INVITE
100 Trying
200 OK (SUBSCRIBE)
183 Session Progress
183 Session Progress
PRACK
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 4: Calls to Lync Clients Fail
144
When a user dials a Lync client from a video-enabled Cisco 9951 phone, the call fails. Call is from 58574 to 60051.
Case Study 4: Live Demo
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Case Study 4: Calls to Lync Clients Fail
146
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
© 2013 Cisco and/or its affiliates. All rights reserved. BRKUCC-2932 Cisco Public
Case Study 4: Calls to Lync Clients Fail
For more information:
– Visit http://developer.cisco.com/web/sip/documentation to download the SIP Normalization and Transparency Developer Guide
147
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