troubleshooting sip with cisco unified...

Troubleshooting SIP with Cisco Unified Communications BRKUCC-2932

Paul Giralt

Distinguished Services Engineer

[email protected]

© 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


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)

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/2.0 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

10


SIP Request Line


SIP Method

URI SIP Version

11


SIP Headers


12


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

13


SIP Response


Response Code

Free-text Reason

14


SIP Response


15


SIP Responses

Response Code Description Example

1xx Informational – Request Received and Continuing to Process

Request

100 Trying

180 Ringing

183 Session Progress

2xx Success – Action was successfully received, understood, and

accepted

200 OK

202 Acceptable

3xx Redirection – Another SIP Element needs to be contacted in order

to complete the request

300 Multiple Choices

301 Moved Permanently

302 Moved Temporarily

4xx Client Error – Request contains bad syntax or cannot be fulfilled at

this server

401 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

16


INVITE

200 OK

Session Established

Phone 1 Unified CM

Basic SIP Call Setup

ACK

BYE

200 OK

17


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)

18


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


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) - 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


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 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.


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


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


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


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


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


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


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


DTMF Relay

3 Methods for passing DTMF digits over a SIP network:

– RFC 2833

– SIP NOTIFY

– SIP Keypad Markup Language (KPML)

37


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=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 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



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



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


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 SIP KPML

42

Passes DTMF information in a SIP NOTIFY message kpml Event

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


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 Relay SIP 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 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


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 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


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

Wireshark

47


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.


RTMT Session Trace Tool

49



Call Flow Diagram

50



Click on the message in the call flow diagram to see the actual message

51


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)

http://translatorx.cisco.com/


TranslatorX Tool

53


TranslatorX Tool Call List Window

54


TranslatorX Tool Call List Filtering

55

Double-click for complete

Call Detail Record


TranslatorX Tool CDR View

56


TranslatorX Tool CDR View

57


TranslatorX Tool Call List Filtering

58

Select a Call and click

“Generate Filter” button


TranslatorX Tool Message 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


Wireshark VoIP Call Analysis

64


Wireshark VoIP Call Ladder Diagram

65


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


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



72

Can Also Use the Troubleshooting Trace Settings Page in CallManager Serviceability (Trace > Troubleshooting Trace Settings)


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


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

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Cisco Unified Border Element (CUBE) Tracing Configuration


CUBE Debugging

CUBE / IOS Tools:

– IOS debugs

– IOS show commands

– Per-call trace

– Packet export

76


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


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


CUBE Debugging Sample ‘debug ccsip messages’

79

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)

80


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


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


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


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


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


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.


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


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


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:


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 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


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 #2 Problem Description

92

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 #2 Use RTMT Session Trace

93

Enter *5551212 into Called Number/URI field

Set time and duration appropriately

Search Finds two calls


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


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


sip

voice service voip

ip address trusted list

ipv4 172.18.106.0 255.255.255.0


sip

OR

<- PREFERRED (More Secure)


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


Case Study 3: No One Answers the Phone Collect Traces

97

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


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.


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

99

Search for called party number



100

Disable Filters

Select the INVITE

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



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



102

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



103

Select the INVITE

Create New Filter (control/command-N)

Filter by IP Address (control/command – I)

Re-enable Filters



104


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


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: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 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


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


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>


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>


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

111


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

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


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"/>


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


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



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>


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



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)


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"/>


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


Case Study 3: No One Answers the Phone User Dials Remaining Digits

121



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


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



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]


|PatternPrecedenceLevel=PlDefault

|PatternRouteClass=RouteClassDefault

|RouteNextHopByCgpn=false

>

}

}


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



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


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


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


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



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


133


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


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

135


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 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


CSeq: 101 CANCEL

Content-Length: 0



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


183 Session Progress (w/ ANSWER)

Phone 1


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)



Phone 1


Unified CM

CUBE

SBC (CUBE) SIP

SP SBC

SP SBC

139


INVITE (w/ OFFER)

183 Session Progress (w/ OFFER)

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



Phone 1


Unified CM

CUBE

SBC (CUBE) SIP

SP SBC

SP SBC

140


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

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


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

142



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)



PRACK


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



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



For more information:

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

147

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


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