Troubleshooting SIP with Cisco Unified Communicationsd2zmdbbm9feqrf.cloudfront.net/2015/eur/pdf/BRKUCC-2932.pdf · Troubleshooting SIP with Cisco Unified Communications BRKUCC-2932

Troubleshooting SIP with Cisco Unified Communications

BRKUCC-2932

Paul Giralt (@PaulGiralt)

Distinguished Services Engineer

[email protected]

© 2014 Cisco and/or its affiliates. All rights reserved.BRKUCC-2392 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


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


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

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

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

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

9


SIP INVITE Method

INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK1515b3154665From: "Test User 1" <sip:[email protected]>;tag=97903bc0-43adcd-45510543To: <sip:[email protected]>Call-ID: [email protected]: timer,resource-priority,replacesUser-Agent: Cisco-CUCM8.6Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFERCSeq: 101 INVITEExpires: 180Allow-Events: presence, kpmlSupported: X-cisco-srtp-fallbackSupported: GeolocationCall-Info: <sip:172.18.106.59:5060>;method="NOTIFY;Event=telephone-event;Duration=500"Cisco-Guid: 2081204224-3137452793-0000000466-0996807340Session-Expires: 1800P-Asserted-Identity: "Test User 1" <sip:[email protected]>Contact: <sip:[email protected]:5060>;video;audioMax-Forwards: 69Content-Length: 864Content-Type: application/sdp

10


SIP Request Line


11

SIP Method

URI SIP Version


SIP Headers


12


SIP Response

SIP/2.0 404 Not FoundVia: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bKb5291d44b969a4From: "Paul Giralt" <sip:[email protected]>;tag=19210123ca7-45568313To: <sip:[email protected]>;tag=253488-726Date: Mon, 16 Jan 2012 04:00:22 GMTCall-ID: [email protected]: 101 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.2.TReason: Q.850;cause=1Content-Length: 0

13


SIP Response


14

Response Code

Free-text Reason


SIP Response


15


SIP Responses

16

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


INVITE

200 OK

Session Established

Phone 1 Unified CM

Basic SIP Call Setup

17

ACK

BYE

200 OK


INVITE

200 OK

Session Established

Phone 1 Unified CM

Basic SIP Call Setup with B2BUA (Unified CM)

18

ACK

BYE

200 OK

Phone 2

INVITE

200 OK

ACK

BYE

200 OK

CUBE

SBC (CUBE)


INVITE

200 OK

Phone 1

Basic SIP Call Setup with Unified CM and CUBE

19

ACK

BYE

200 OK

Unified CM

CUBE

SBC (CUBE)

INVITE

200 OK

ACK

BYE

200 OK

INVITE

200 OK

ACK

BYE

200 OK

SIP

SPSBC

SP SBC

Session Established


Media Negotiation

• SIP leverages the Session Description Protocol (SDP) (RFC 4566/3266/2327) to communicate media information.

• SIP uses the offer/answer model described in RFC 3264 to negotiate media using SDP

20


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.”

21


Session Description Protocol (SDP) - Offerv=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=recvonly22


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=ptime:20


a=fmtp:101 0-15


23


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.

24


INVITE with SDP - Offer

200 OK with SDP - Answer

Session Established

Phone 1 Unified CM

Early Offer

25

ACK (no SDP)

BYE

200 OK


INVITE (no SDP)

200 OK with SDP - Offer

Session Established

Phone 1 Unified CM

Delayed Offer

26

ACK with SDP - Answer

BYE

200 OK


Early Media

• Delayed Offer calls do not set up media until the 200 OK (call is answered)

• If media is required prior to the call being connected, SIP has provisions for Early Media

• With Early Media on a Delayed Offer call, the offer comes from the terminating side in a provisional response (e.g. 183 Session Progress)

• Originating side sends SDP Answer in a PRACK message (defined in RFC 3262)

27


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)


Media Re-negotiation

• 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.

Re-INVITE

29



INVITE sip:[email protected]:49833;transport=tls SIP/2.0Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK901f9c72c19221From: "Paul Giralt" <sip:[email protected]>;tag=15462272~0d0d25d7-4931-4a07-83c6-b82e2c213ca7-45545776To: <sip:[email protected]>;tag=0022bdd6843100702aae8e5b-4be253beDate: Wed, 11 Jan 2012 03:08:51 GMTCall-ID: [email protected]: timer,resource-priority,replacesMin-SE: 1800User-Agent: Cisco-CUCM8.6Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 104 INVITEMax-Forwards: 70Expires: 180Allow-Events: presenceCall-Info: <urn:x-cisco-remotecc:callinfo>; security= Authenticated; orientation= from; gci= 2-231448; call-instance= 2Remote-Party-ID: "Paul Giralt" <sip:[email protected]>;party=calling;screen=yes;privacy=offContact: <sip:[email protected]:5061;transport=tls>Content-Type: application/sdpContent-Length: 489

Re-INVITE

30



v=0o=CiscoSystemsCCM-SIP 15462272 2 IN IP4 172.18.106.59s=SIP Callc=IN IP4 0.0.0.0t=0 0m=audio 19594 RTP/SAVP 9 101a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXa=rtpmap:9 G722/8000a=ptime:20a=inactivea=rtpmap:101 telephone-event/8000a=fmtp:101 0-15m=video 19444 RTP/AVP 126b=TIAS:1000000a=rtpmap:126 H264/90000a=fmtp:126 profile-level-id=42801E;packetization-mode=1;level-asymmetry-allowed=1a=inactivea=mid:227796888

Re-INVITE – Stopping a Media Session

31



INVITE sip:[email protected]:49833;transport=tls SIP/2.0Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK901fac34c0fb1bFrom: "Paul Giralt" <sip:[email protected]>;tag=15462272~0d0d25d7-4931-4a07-83c6-b82e2c213ca7-45545776To: <sip:[email protected]>;tag=0022bdd6843100702aae8e5b-4be253beDate: Wed, 11 Jan 2012 03:08:52 GMTCall-ID: [email protected]: timer,resource-priority,replacesMin-SE: 1800User-Agent: Cisco-CUCM8.6Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 106 INVITEMax-Forwards: 70Expires: 180Allow-Events: presenceCall-Info: <urn:x-cisco-remotecc:callinfo>; security= NotAuthenticated; orientation= from; gci= 2-231448; call-instance= 2Remote-Party-ID: "Paul Giralt" <sip:[email protected]>;party=calling;screen=yes;privacy=offContact: <sip:[email protected]:5061;transport=tls>Content-Length: 0

Re-INVITE – Delayed Offer to Re-establish Media Stream

32



SIP/2.0 200 OKVia: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK901fac34c0fb1bFrom: "Paul Giralt" <sip:[email protected]>;tag=15462272~0d0d25d7-4931-4a07-83c6-b82e2c213ca7-45545776To: <sip:[email protected]>;tag=0022bdd6843100702aae8e5b-4be253beCall-ID: [email protected]: Wed, 11 Jan 2012 03:08:52 GMTCSeq: 106 INVITEServer: Cisco-CPCIUS/9.2.1Contact: <sip:[email protected]:49833;transport=tls>Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFORemote-Party-ID: "Paul Giralt" <sip:[email protected]>;party=called;id-type=subscriber;privacy=off;screen=yesSupported: 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-8.5.1Allow-Events: kpml,dialogRecv-Info: conferenceRecv-Info: x-cisco-conferenceContent-Length: 788Content-Type: application/sdpContent-Disposition: session;handling=optional

Re-INVITE – Offer in 200 OK

33



v=0o=Cisco-SIPUA 26259 2 IN IP4 10.116.101.41s=SIP Callt=0 0m=audio 32518 RTP/SAVP 0 8 18 102 9 116 124 101c=IN IP4 10.116.101.41a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXa=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=rtpmap:102 L16/16000a=rtpmap:9 G722/8000a=rtpmap:116 iLBC/8000a=rtpmap:124 ISAC/16000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=sendrecvm=video 17614 RTP/AVP 126 97c=IN IP4 10.116.101.41b=TIAS:2500000a=rtpmap:126 H264/90000a=fmtp:126 profile-level-id=42801F;packetization-mode=1;level-asymmetry-allowed=1a=rtpmap:97 H264/90000a=fmtp:97 profile-level-id=42801F;packetization-mode=0;level-asymmetry-allowed=1a=sendrecv

Re-INVITE – Offer in 200 OK

34



ACK sip:[email protected]:49833;transport=tls SIP/2.0Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK901fb064465a06From: "Paul Giralt" <sip:[email protected]>;tag=15462272~0d0d25d7-4931-4a07-83c6-b82e2c213ca7-45545776To: <sip:[email protected]>;tag=0022bdd6843100702aae8e5b-4be253beDate: Wed, 11 Jan 2012 03:08:52 GMTCall-ID: [email protected]: 70CSeq: 106 ACKAllow-Events: presenceContent-Type: application/sdpContent-Length: 446

Re-INVITE – Answer in ACK

35



v=0o=CiscoSystemsCCM-SIP 15462272 3 IN IP4 172.18.106.59s=SIP Callt=0 0m=audio 4000 RTP/SAVP 0c=IN IP4 172.18.106.58a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXa=rtpmap:0 PCMU/8000a=ptime:20a=sendonlym=video 0 RTP/AVP 126c=IN IP4 10.116.101.50b=TIAS:1000000a=rtpmap:126 H264/90000a=fmtp:126 profile-level-id=42801E;packetization-mode=1;level-asymmetry-allowed=1a=mid:227796888

Re-INVITE – Answer in ACK – Decline Video Support

36


DTMF Relay

• 3 Methods for passing DTMF digits over a SIP network:

– RFC 2833

– SIP NOTIFY

– SIP Keypad Markup Language (KPML)

37


DTMF Relay

• Digits are passed in the RTP stream with a unique payload type

• Capability is negotiated in SDP like any other codec

RFC 2833

38

m=audio 30414 RTP/AVP 0 8 116 18 100 101

c=IN IP4 172.18.106.231




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=fmtp:101 0-16

m=audio 17236 RTP/AVP 0 101


a=ptime:20


a=fmtp:101 0-15

Offer Answer


DTMF Relay

• Passes DTMF information in a SIP NOTIFY message telephone-event Event

• Negotiated in Call-Info header

SIP NOTIFY

39

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


DTMF RelaySIP NOTIFY

40

SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK9843c455840434


To: <sip:[email protected]>;tag=4363A830-17FC


... snip ...

Allow-Events: telephone-event

Call-Info: <sip:172.18.106.231:5060>;method="NOTIFY;Event=telephone-event;Duration=500”

... snip ...

Content-Length: 601

Answer


DTMF Relay

• Digits passed in payload of a NOTIFY message

SIP NOTIFY

41

NOTIFY sip:172.18.106.231:5060 SIP/2.0

Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK98443140152a0a


To: <sip:[email protected]>;tag=4363A830-17FC


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


DTMF Relay

• Passes DTMF information in a SIP NOTIFY message kpml Event

• Capability advertised in Allow-Events – uses SUBSCRIBE message to subscribe

SIP KPML

42


Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK986efd6c4e51e4



Date: Mon, 13 May 2013 15:05:24 GMT



... snip ...

Allow-Events: presence, kpml

... snip ...

Session-Expires: 18000

Max-Forwards: 69

Content-Length: 0

Offer


DTMF RelaySIP KPML

43

SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK986efd6c4e51e4


To: <sip:[email protected]>;tag=437394E8-2E1

Date: Mon, 13 May 2013 15:05:26 GMT


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


DTMF RelaySIP 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


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


DTMF RelaySIP KPML

45

NOTIFY sip:172.18.106.231:5060 SIP/2.0

Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK986f73662cca3b


To: <sip:[email protected]>;tag=437394E8-2E1


CSeq: 104 NOTIFY

Max-Forwards: 70


Event: kpml

Subscription-State: active;expires=7197


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


SIP Troubleshooting Tools

• Unified CM / SME Tools:

– Real Time Monitoring Tool / Session Trace

– TranslatorX

• IOS (CUBE) and VCS Troubleshooting Tool:

– TranslatorX

• Wireshark

47


RTMT Session Trace Tool

• 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.

Session Trace Features

48


RTMT Session Trace Tool

49


RTMT Session Trace ToolCall Flow Diagram

50


RTMT Session Trace ToolClick on the message in the call flow diagram to see the actual message

51


TranslatorX Tool

• 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, event-trace, and VCS diagnostic log 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

• 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)

Features

52

http://translatorx.cisco.com/


TranslatorX Tool

53


TranslatorX ToolCall List Window

54


TranslatorX ToolCall List Filtering

55

Double-click for complete

Call Detail Record


TranslatorX ToolCDR View

56


TranslatorX ToolCDR View

57


TranslatorX ToolCall List Filtering

58

Select a Call and click

“Generate Filter” button


TranslatorX ToolMessage Filters

59


TranslatorX Tool

60


TranslatorX Tool

61


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

http://www.wireshark.org


Wireshark

63


WiresharkVoIP Call Analysis

64


WiresharkVoIP Call Ladder Diagram

65


WiresharkHow to Gather a Trace?

66

• Unified CM, VCS, and IOS provide a mechanism to gather a packet capture

• Will be covered later

Unified CM Tracing Configuration


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



69

Select the

Server

Select the Service on

Which Trace Needs to

Be Enabled

Select Service

Group



70

1. Press

Set Default

2. Set to Detailed

Updates All

Servers in This

Cluster with

These Settings



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.

Okay to have enabled in 9.x and later


Trace Collection• Various Ways to Collect Trace Files

72

• RTMT Analysis Manager

• RTMT Remote Browse

• RTMT Collect Files

• RTMT Query Wizard

• OS CLI (file get or file tail)

– file tail activelog cm/trace/ccm/sdl recent

Recommended


Gathering a Packet Capture from Unified CM

• Use the Platform CLI command ‘utils network capture’

73

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

VCS Tracing Configuration


VCS Trace Configuration

• VCS Control / VCS Expressway / Expressway-C / Expressway-E can log SIP messages to a diagnostic log file.

• To enable, navigate to Maintenance > Diagnostics > Diagnostic logging

75



76

Select this if you want

to get a Wireshark

capture

Click Start new log

Click OK to Confirm



77

Click Stop logging after

reproducing your problem



78

Click to Download the log or

tcpdump (for Wireshark)



79

Other trace settings can

be enabled here

Cisco Unified Border Element (CUBE) Tracing Configuration


CUBE Debugging

• CUBE / IOS Tools:– IOS debugs

– IOS show commands

– Per-call trace

– Event Trace

– Packet export

81


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

82

service timestamps debug datetime msec localtime

service timestamps log datetime msec localtime

logging buffered 10000000

no logging console

clock timezone EST -5 0

clock summer-time EDT recurring

ntp server 10.14.1.1


CUBE Debugging

• Various SIP debugs available:

83

CUBE#debug ccsip ?

all Enable all SIP debugging traces

calls Enable CCSIP SPI calls debugging trace

dhcp Enable SIP-DHCP debugging trace

error Enable SIP error debugging trace

events Enable SIP events debugging trace

function Enable SIP function debugging trace

info Enable SIP info debugging trace

media Enable SIP media debugging trace

messages Enable CCSIP SPI messages debugging trace

preauth Enable SIP preauth debugging traces

states Enable CCSIP SPI states debugging trace

translate Enable SIP translation debugging trace

transport Enable SIP transport debugging traces

verbose Enable verbose mode


CUBE DebuggingSample ‘debug ccsip messages’

84

Jan 12 03:14:43.102: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:


Via: SIP/2.0/TCP 172.18.106.59:5060;branch=z9hG4bK978d2e8df73dc

From: "Paul Giralt" <sip:[email protected]>;tag=16218435~-b82e2c213ca7-45552048


Date: Thu, 12 Jan 2012 03:09:42 GMT


Supported: 100rel,timer,resource-priority,replaces

Min-SE: 1800


Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY

CSeq: 101 INVITE

Expires: 180


Supported: X-cisco-srtp-fallback

Supported: Geolocation


Cisco-Guid: 3720668288-0000065536-0000015564-0996807340



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


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)

85


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

86

Incoming VoIP Call Outgoing VoIP Call

dial-peer voice 1 voipdestination-pattern 1000incoming called-number .Tsession protocol sipv2session target ipv4:192.168.10.50dtmf-relay rtp-nte sip-kpmlcodec g711ulaw

dial-peer voice 2 voipdestination-pattern 2000session protocol sipv2session target ipv4:192.168.12.25dtmf-relay rtp-ntecodec g711ulaw

Originating Endpoint

TerminatingEndpoint

CUBE

voice service voip

allow-connections sip to sip


CUBE show Commands

• show call active voice [brief] shows state of currently active calls

87

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


CUBE show Commands

• show cube calls shows more specific CUBE-related call information and allows filtering

88

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


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

89


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 contentper-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

90

Reference


CUBE SIP Event Trace

• All events related to a call are put into buffers

• Can search for calls based on calling number, called number, or call ID

• Buffers can be automatically dumped to an FTP/TFTP server when calls end

• Available on CUBE(Ent) ASR release XE3.10 and later

• Available on CUBE(Ent) ISR release 15.3(3)M and later

91


1. Define events to be traced

2. Enable event trace (will enable automatically on a reload)

CUBE(config)# monitor event-trace voip ccsip msg size 50

CUBE SIP Event Trace

92

CUBE# monitor event-trace voip ccsip msg ?

clear Clear the trace

disable Disable tracing

dump Dumps the event buffer into a file

enable Enable tracing

CUBE(config)# monitor event-trace voip ccsip dump-file ftp://user:password@<ipaddr>/path/cube.txt

CUBE(config)# monitor event-trace voip ccsip dump all

3. Configure automatic file uploads (optional)


CUBE SIP Event TraceViewing event trace information

93

CUBE# monitor event-trace voip ccsip [msg | history] dump filter ? [pretty]

call-id Filter traces based on Internal Call Id

called-num Filter traces based on Called Number

calling-num Filter traces based on Calling Number

sip-call-id Filter traces based on SIP Call Id

CUBE# show monitor event-trace voip ccsip [msg | history] ?

all Show all the traces in current buffer

back Show trace from this far back in the past

clock Show trace from a specific clock time/date

filter Show Trace filter Options

from-boot Show trace from this many seconds after booting

latest Show latest trace events since last display


CUBE SIP Event TraceViewing event trace information

94

CUBE# show monitor event-trace voip ccsip history filter calling-num 9195551234 latest

--------Cover buff----------

buffer-id = 2828 ccCallId = 305319 PeerCallId = 305320

Called-Number = +18045553456 Calling-Number = 9195551234 Sip-Call-Id

= [email protected]

sip_msgs: Enabled.. Total Traces logged = 18

--------------------------------

--------Cover buff----------

buffer-id = 2829 ccCallId = 305320 PeerCallId = 305319

Called-Number = 8045553456 Calling-Number = 9195551234 Sip-Call-Id =

[email protected]

sip_msgs: Enabled.. Total Traces logged = 12

--------------------------------


CUBE – IP Traffic CaptureExport Packet Data in PCAP Format

95

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


Case Study 1: Unable to place a callProblem Description

97

• A user reports that every time they call (919) 555-1212, they get a message that the call could not be completed as dialed.


Case Study 1: Unable to Place a CallUse RTMT Session Trace

98

• Enter *5551212 into Called Number/URI field

• Set time and duration appropriately

• Search Finds two calls


Case Study 1: Unable to Place a Call

• Double-click to see message diagram

• Clearly shows the far-end sends back a 404 Not Found

Use RTMT Session Trace

99


Case Study 1: Unable to Place a Call

• Enable SIP message debugs – debug ccsip messages

Troubleshoot Call on CUBE

100

Jan 16 04:00:22.679: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:


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


Date: Mon, 16 Jan 2012 03:55:17 GMT


Supported: timer,resource-priority,replaces

Min-SE: 1800


Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY

CSeq: 101 INVITE

Expires: 180



Cisco-Guid: 3852191232-0000065536-0000018595-0996807340




Max-Forwards: 69

Content-Length: 0


Case Study 1: Unable to Place a CallTroubleshoot Call on CUBE

101

• 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


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


Case Study 2: Unable to Place a Call #2Problem Description

102

• A user reports that every time they call (919) 555-1212, they get reorder (fast busy) tone.


Case Study 2: Unable to Place a Call #2Use RTMT Session Trace

103

• Enter *5551212 into Called Number/URI field

• Set time and duration appropriately

• Search Finds two calls


Case Study 2: Unable to Place a Call #2Use RTMT Session Trace

104

• Trace shows signaling from phone as well as to CUBE

• CUBE is responding with a 403 Forbidden


Case Study 2: Unable to Place a Call #2Problem Description

105

• 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


sip

voice service voip

ip address trusted list

ipv4 172.18.106.0 255.255.255.0


sip

OR

<- RECOMMENDED (More Secure)


Case Study 3: No One Answers the PhoneProblem Description

106

• 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


Case Study 3: No One Answers the PhoneCollect Traces

107

• Problem is reproducible, so generate a test call and then collect traces.


Case Study 3: No One Answers the PhoneUse TranslatorX

108

• 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.


Case Study 3: No One Answers the PhoneUse TranslatorX to Analyze Traces

109

• Search for called party number



110

• Disable Filters

• Select the INVITE

• Filter by SIP Call ID (control/command – S)



111

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/2.0Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK1515b3154665From: "Test User 1" <sip:[email protected]>;tag=97903bc0-a3de-4a15-ba27-44c81fe3adcd-45510543To: <sip:[email protected]>Date: Mon, 29 Mar 2010 14:36:41 GMTCall-ID: [email protected]: timer,resource-priority,replacesMin-SE: 1800User-Agent: Cisco-CUCM8.0Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 101 INVITEExpires: 180Allow-Events: presence, kpmlSupported: X-cisco-srtp-fallbackSupported: GeolocationCall-Info: <sip:172.18.106.59:5060>;method="NOTIFY;Event=telephone-event;Duration=500"Cisco-Guid: 2081204224-3137452793-0000000466-0996807340Session-Expires: 1800P-Asserted-Identity: "Test User 1" <sip:[email protected]>Contact: <sip:[email protected]:5060>;video;audioMax-Forwards: 69Content-Length: 0



112

• Where did the call originate? Try searching for the calling party number



113

• Select the INVITE

• Create New Filter (control/command-N)

• Filter by IP Address (control/command – I)

• Re-enable Filters



114


Case Study 3: No One Answers the PhoneINVITE from IP Phone w/ SDP

115

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/2.0Via: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK1636ab61From: "Test User 1" <sip:[email protected]>;tag=00260bd9669e07147bcb3aac-3cda8f0cTo: <sip:[email protected];user=phone>Call-ID: [email protected]: 70Date: Mon, 29 Mar 2010 14:36:33 GMTCSeq: 101 INVITEUser-Agent: Cisco-CP9951/9.0.1Contact: <sip:[email protected]:51682;transport=tls>Expires: 180Accept: application/sdpAllow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFORemote-Party-ID: "Test User 1" <sip:[email protected]>;party=calling;id-type=subscriber;privacy=off;screen=yesSupported: 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-9.0.1Allow-Events: kpml,dialogContent-Length: 632Content-Type: application/sdpContent-Disposition: session;handling=optional


Case Study 3: No One Answers the PhoneINVITE from IP Phone w/ SDP (continued)

116

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:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:102 L16/16000

a=rtpmap:9 G722/8000


a=fmtp:116 mode=20

a=rtpmap:124 ISAC/16000


a=fmtp:101 0-15

a=sendrecv


c=IN IP4 172.18.159.152

b=TIAS:1000000

a=rtpmap:97 H264/90000

a=fmtp:97 profile-level-id=42801E

a=recvonly


Case Study 3: No One Answers the PhoneUnified CM Sends a 100 Trying

117

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 TryingVia: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK1636ab61From: "Test User 1" <sip:[email protected]>;tag=00260bd9669e07147bcb3aac-3cda8f0cTo: <sip:[email protected];user=phone>Date: Mon, 29 Mar 2010 14:36:33 GMTCall-ID: [email protected]: 101 INVITEAllow-Events: presenceContent-Length: 0


Case Study 3: No One Answers the PhoneUnified CM Sends a REFER to Play Outside Dialtone

118

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/2.0Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK151511c5f04bfFrom: <sip:[email protected]>;tag=2144536187To: <sip:[email protected]>Call-ID: [email protected]: 101 REFERMax-Forwards: 70Contact: <sip:[email protected]:5061;transport=tls>User-Agent: Cisco-CUCM8.0Expires: 0Refer-To: cid:[email protected]: <[email protected]>Require: norefersubContent-Type: application/x-cisco-remotecc-request+xmlReferred-By: <sip:[email protected]>Content-Length: 409


Case Study 3: No One Answers the PhoneUnified CM Sends a REFER to play Outside Dialtone (continued)

119

<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>


Case Study 3: No One Answers the PhoneUnified CM Sends a SUBSCRIBE for KPML

120

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/2.0Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK1515232b4e84fFrom: <sip:[email protected]>;tag=1976165806To: <sip:[email protected]>Call-ID: [email protected]: 101 SUBSCRIBEDate: Mon, 29 Mar 2010 14:36:33 GMTUser-Agent: Cisco-CUCM8.0Event: kpml; [email protected]; from-tag=00260bd9669e07147bcb3aac-3cda8f0cExpires: 7200Contact: <sip:[email protected]:5061;transport=tls>Accept: application/kpml-response+xmlMax-Forwards: 70Content-Type: application/kpml-request+xmlContent-Length: 424<?xml version="1.0" encoding="UTF-8" ?><kpml-request xmlns="urn:ietf:params:xml:ns:kpml-request" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"

xsi:schemaLocation="urn:ietf:params:xml:ns:kpml-request kpml-request.xsd" version="1.0"><pattern criticaldigittimer="1000" extradigittimer="500" interdigittimer="10000" persist="persist"><regex tag="Backspace OK">[x#*+]|bs</regex>

</pattern></kpml-request>


Case Study 3: No One Answers the PhonePhone Sends 200 OK for the REFER and SUBSCRIBE

121


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


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


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


Case Study 3: No One Answers the Phone

122

IP Phone(172.18.159.152)

Unified CM(172.18.159.152)

CUBE(172.18.159.231)

INVITE

100 Trying

REFER

SUBSCRIBE

200 OK (REFER)

200 OK (SUBSCRIBE)


Case Study 3: No One Answers the PhoneUser Dials a ‘1’

123


NOTIFY sip:[email protected]:5061 SIP/2.0Via: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK1cd529baTo: <sip:[email protected]>;tag=1976165806From: <sip:[email protected]>;tag=00260bd9669e07177ee0d51d-14f56f89Call-ID: [email protected]: Mon, 29 Mar 2010 14:36:33 GMTCSeq: 1001 NOTIFYEvent: kpmlSubscription-State: active; expires=7200Max-Forwards: 70Contact: <sip:[email protected]:51682;transport=TLS>Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBEContent-Length: 209Content-Type: application/kpml-response+xmlContent-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"/>


Case Study 3: No One Answers the PhoneUnified CM Replies to NOTIFY With a 200 OK

124

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 OKVia: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK1cd529baFrom: <sip:[email protected]>;tag=00260bd9669e07177ee0d51d-14f56f89To: <sip:[email protected]>;tag=1976165806Date: Mon, 29 Mar 2010 14:36:34 GMTCall-ID: [email protected]: 1001 NOTIFYContent-Length: 0


Case Study 3: No One Answers the PhoneUnified CM Replies Sends a REFER to Disable Outside Dialtone

125

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:[email protected]:51682 SIP/2.0Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK151536ea86ab0From: <sip:[email protected]>;tag=1574166193To: <sip:[email protected]>Call-ID: [email protected]: 101 REFERMax-Forwards: 70Contact: <sip:[email protected]:5061;transport=tls>User-Agent: Cisco-CUCM8.0Expires: 0Refer-To: cid:[email protected]: <[email protected]>Require: norefersubContent-Type: application/x-cisco-remotecc-request+xmlReferred-By: <sip:[email protected]>Content-Length: 401



126

<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>


Case Study 3: No One Answers the PhonePhone Replies With 200 OK to REFER

127


SIP/2.0 200 OKVia: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK151536ea86ab0From: <sip:[email protected]>;tag=1574166193To: <sip:[email protected]>;tag=00260bd9669e07184b08b96b-796ab86fCall-ID: [email protected]: Mon, 29 Mar 2010 14:36:33 GMTCSeq: 101 REFERServer: Cisco-CP9951/9.0.1Contact: <sip:[email protected]:51682;transport=TLS>Content-Length: 0



128

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)


Case Study 3: No One Answers the PhoneUser Dials a ‘8’

129


NOTIFY sip:[email protected]:5061 SIP/2.0Via: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK647d03c1To: <sip:[email protected]>;tag=1976165806From: <sip:[email protected]>;tag=00260bd9669e07177ee0d51d-14f56f89Call-ID: [email protected]: Mon, 29 Mar 2010 14:36:34 GMTCSeq: 1002 NOTIFYEvent: kpmlSubscription-State: active; expires=7195Max-Forwards: 70Contact: <sip:[email protected]:51682;transport=TLS>Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBEContent-Length: 209Content-Type: application/kpml-response+xmlContent-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"/>


Case Study 3: No One Answers the PhoneUnified CM Replies to NOTIFY With a 200 OK

130


SIP/2.0 200 OKVia: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK1cd529baFrom: <sip:[email protected]>;tag=00260bd9669e07177ee0d51d-14f56f89To: <sip:[email protected]>;tag=1976165806Date: Mon, 29 Mar 2010 14:36:34 GMTCall-ID: [email protected]: 1001 NOTIFYContent-Length: 0


Case Study 3: No One Answers the PhoneUser Dials Remaining Digits

131



132

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


Case Study 3: No One Answers the PhoneDigit Analysis Match

133

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



134

|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



135

|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]


|PatternPrecedenceLevel=PlDefault

|PatternRouteClass=RouteClassDefault

|RouteNextHopByCgpn=false

>

}

}



136

|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=



137

|UnicodeDisplayName=

|DisplayNameLocale=1

|OverlapSendingFlagEnabled=0

|WithTags=

|WithValues=

|CallingPartyNumberPi=NotSelected

|ConnectedPartyNumberPi=NotSelected

|CallingPartyNamePi=NotSelected

|ConnectedPartyNamePi=NotSelected


|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


Case Study 3: No One Answers the PhoneCUCM Sends an INVITE to the CUBE

138

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/2.0Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK1515b3154665From: "Test User 1" <sip:[email protected]>;tag=97903bc0-a3de-4a15-ba27-44c81fe3adcd-45510543To: <sip:[email protected]>Date: Mon, 29 Mar 2010 14:36:41 GMTCall-ID: [email protected]: timer,resource-priority,replacesMin-SE: 1800User-Agent: Cisco-CUCM8.0Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 101 INVITEExpires: 180Allow-Events: presence, kpmlSupported: X-cisco-srtp-fallbackSupported: GeolocationCall-Info: <sip:172.18.106.59:5060>;method="NOTIFY;Event=telephone-event;Duration=500"Cisco-Guid: 2081204224-3137452793-0000000466-0996807340Session-Expires: 1800P-Asserted-Identity: "Test User 1" <sip:[email protected]>Contact: <sip:[email protected]:5060>;video;audioMax-Forwards: 69Content-Length: 0



139

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

INVITE200 OK (SUBSCRIBE)


Case Study 3: No One Answers the PhoneCUBE Replies With a 183 Session Progress W/ SDP

140

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 ProgressVia: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK1515b3154665From: "Test User 1" <sip:[email protected]>;tag=97903bc0-a3de-4a15-ba27-44c81fe3adcd-45510543To: <sip:[email protected]>;tag=DE1EFF8-0Date: Mon, 29 Mar 2010 14:37:23 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventRemote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=offContact: <sip:[email protected]:5060>Supported: sdp-anatServer: Cisco-SIPGateway/IOS-12.xContent-Type: multipart/mixed;boundary=uniqueBoundaryMime-Version: 1.0Content-Length: 788--uniqueBoundary


Case Study 3: No One Answers the PhoneCUBE Replies With a 183 Session Progress W/ SDP

141

Content-Type: application/sdpContent-Disposition: session;handling=requiredv=0o=CiscoSystemsSIP-GW-UserAgent 0 7954 IN IP4 172.18.159.231s=SIP Callc=IN IP4 172.18.159.231t=0 0m=audio 27980 RTP/AVP 0 8 116 18 100 101c=IN IP4 172.18.159.231a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:116 iLBC/8000a=fmtp:116 mode=20a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:100 X-NSE/8000a=fmtp:100 192-194a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16--uniqueBoundaryContent-Type: application/x-q931Content-Disposition: signal;handling=optionalContent-Length: 11


Case Study 3: No One Answers the PhoneUnified CM Sends a 180 Ringing to the IP Phone

142


SIP/2.0 180 RingingVia: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK1636ab61From: "Test User 1" <sip:[email protected]>;tag=00260bd9669e07147bcb3aac-3cda8f0cTo: <sip:[email protected];user=phone>;tag=97903bc0-a3de-4a15-ba27-44c81fe3adcd-45510542Date: Mon, 29 Mar 2010 14:36:33 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYAllow-Events: presenceContact: <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= 1Send-Info: conferenceRemote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=offContent-Length: 0



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)

180 Ringing




• Timestamps Jump from 10:36:42 to 10:37:32

• No SIP Signaling for 50 seconds

Phone Keeps Ringing

144


Case Study 3: No One Answers the PhonePhone Sends a CANCEL

145

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>


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


Case Study 3: No One Answers the PhoneUnified CM Sends a 200 OK for the CANCEL

146

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


CSeq: 101 CANCEL

Content-Length: 0



147

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


183 Session Progress (w/ ANSWER)

Phone 1


148

Unified CM

CUBE

SBC (CUBE)SIP

SPSBC

SP SBC

INVITE w/ OFFERINVITE (no SDP)

INVITE (w/ OFFER)

183 Session Progress (w/ OFFER) 180 Ringing (no SDP)



Phone 1


149

Unified CM

CUBE

SBC (CUBE)SIP

SPSBC

SP SBC


INVITE (w/ OFFER)

183 Session Progress (w/ OFFER)

??? (w/ ANSWER) ??? (w/ ANSWER)



Phone 1


150

Unified CM

CUBE

SBC (CUBE)SIP

SPSBC

SP SBC


INVITE (w/ OFFER)

183 Session Progress (w/ OFFER)

183 Session Progress (w/ ANSWER) PRACK (w/ ANSWER)



• 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

151

*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


Case Study 3: No One Answers the PhoneUnified CM SIP Profile Configuration

152



153

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)



PRACK


Case Study 4: Calls to Lync Clients Fail

• When a user dials a Lync client from a video-enabled Cisco 9951 phone, the call fails. Call is from 58574 to 60051.

154

Case Study 4: Live Demo



156

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



• For more information:

– Visit http://developer.cisco.com/web/sip/documentation to download the SIP Normalization and Transparency Developer Guide

157

http://developer.cisco.com/web/sip/documentation

Case Study 5: Live Demo


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• Meet the Engineer

• Lunch time Table Topics

• DevNet zone related labs and sessions

• Recommended Reading: for reading material and further resources for this session, please visit www.pearson-books.com/CLMilan2015

159

http://www.pearson-books.com/CLMilan 2015


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160

Documents

Troubleshooting SIP with Cisco Unified Communicationsd2zmdbbm9feqrf.cloudfront.net/2015/eur/pdf/BRKUCC-2932.pdf · Troubleshooting SIP with Cisco Unified Communications BRKUCC-2932