Troubleshooting SIP with Cisco Unified Communicationsd2zmdbbm9feqrf.cloudfront.net/2012/eur/pdf/BRKUCC-2932.pdf · BRKUCC-2932 Troubleshooting SIP with Cisco Unified Communications

BRKUCC-2932

Troubleshooting SIP with Cisco Unified Communications

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© 2012 Cisco and/or its affiliates. All rights reserved. Cisco Public BRKUCC-2932 2

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Agenda

Introduction

Session Initiation Protocol (SIP) Overview

Troubleshooting Tools

Unified CM Tracing

Cisco Unified Border Element (CUBE) Tracing

Sample Call Flows / Case Studies

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


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


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)


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


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


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


SIP INVITE Method









CSeq: 101 INVITE

Expires: 180





Cisco-Guid: 2081204224-3137452793-0000000466-0996807340




Max-Forwards: 69

Content-Length: 864


SIP Method

URI SIP Version


SIP Headers









CSeq: 101 INVITE

Expires: 180





Cisco-Guid: 2081204224-3137452793-0000000466-0996807340




Max-Forwards: 69

Content-Length: 864



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

406 Not Acceptable

486 Busy Here

487 Request Terminated

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

ACK

BYE

200 OK


INVITE

200 OK

Session Established

Phone 1 Unified CM

Basic SIP Call Setup with B2BUA (Unified CM)

ACK

BYE

200 OK

Phone 2

INVITE

200 OK

ACK

BYE

200 OK

CUBE

SBC (CUBE)


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


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


Offer/Answer Model (RFC 3264)

One endpoint sends an offer SDP containing all the capabilities the endpoint wishes to negotiate.

SDP contains m lines for each media stream being negotiated (i.e. audio, video, content channel, etc…)

Receiving endpoint sends an answer SDP that contains the same or a subset of capabilities received in the offer.

Per RFC 3264, “For each "m=" line in the offer, there MUST be a corresponding "m=“ line in the answer. The answer MUST contain exactly the same number of "m=" lines as the offer.”


Session Description Protocol (SDP) - Offer v=0

o=Cisco-SIPUA 26964 0 IN IP4 172.18.159.152

s=SIP Call

t=0 0

m=audio 29254 RTP/SAVP 0 8 18 102 9 116 124 101

c=IN IP4 172.18.159.152

a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:102 L16/16000

a=rtpmap:9 G722/8000

a=rtpmap:116 iLBC/8000

a=fmtp:116 mode=20

a=rtpmap:124 ISAC/16000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

m=video 25466 RTP/AVP 97

c=IN IP4 172.18.159.152

b=TIAS:1000000

a=rtpmap:97 H264/90000

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

a=recvonly


Session Description Protocol (SDP) - Answer v=0

o=CiscoSystemsCCM-SIP 2000 1 IN IP4 172.18.106.59

s=SIP Call

c=IN IP4 172.18.159.152

t=0 0

m=audio 30308 RTP/AVP 0 101


a=ptime:20


a=fmtp:101 0-15



Media Negotiation

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.

For Delayed Offer, the answer is typically sent in the ACK.


INVITE with SDP - Offer

200 OK with SDP - Answer

Session Established

Phone 1 Unified CM

Early Offer

ACK (no SDP)

BYE

200 OK


INVITE (no SDP)

200 OK with SDP - Offer

Session Established

Phone 1 Unified CM

Delayed Offer

ACK with SDP - Answer

BYE

200 OK


Early Media

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

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

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

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


INVITE (no SDP)

200 OK (INVITE) w/ SDP (should be same as answer)

Session Established

Phone 1 Unified CM

Early Media

ACK

BYE

200 OK

183 Session Progress with SDP - Offer

PRACK with SDP - Answer

Media Stream Established

200 OK (PRACK)


Media Re-negotiation

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



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



Min-SE: 1800


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-Length: 489

Re-INVITE



v=0


s=SIP Call

c=IN IP4 0.0.0.0

t=0 0

m=audio 19594 RTP/SAVP 9 101

a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX

XXXXXXXXX


a=ptime:20

a=inactive


a=fmtp:101 0-15


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

Re-INVITE – Stopping a Media Session



INVITE sip:[email protected]:49833;transport=tls SIP/2.0

Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK901fac34c0fb1b


b82e2c213ca7-45545776


Date: Wed, 11 Jan 2012 03:08:52 GMT



Min-SE: 1800



CSeq: 106 INVITE

Max-Forwards: 70

Expires: 180


Call-Info: <urn:x-cisco-remotecc:callinfo>; security= NotAuthenticated; orientation= from; gci= 2-231448;

call-instance= 2



Content-Length: 0

Re-INVITE – Delayed Offer to re-establish media stream



SIP/2.0 200 OK

Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK901fac34c0fb1b


b82e2c213ca7-45545776



Date: Wed, 11 Jan 2012 03:08:52 GMT

CSeq: 106 INVITE

Server: Cisco-CPCIUS/9.2.1


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-Disposition: session;handling=optional

Re-INVITE – Offer in 200 OK



v=0


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:18 G729/8000

a=rtpmap:102 L16/16000





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

Re-INVITE – Offer in 200 OK



ACK sip:[email protected]:49833;transport=tls SIP/2.0

Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK901fb064465a06


b82e2c213ca7-45545776


Date: Wed, 11 Jan 2012 03:08:52 GMT


Max-Forwards: 70

CSeq: 106 ACK



Content-Length: 446

Re-INVITE – Answer in ACK



v=0


s=SIP Call

t=0 0

m=audio 4000 RTP/SAVP 0

c=IN IP4 172.18.106.58

a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX

XXXXXXXXX


a=ptime:20

a=sendonly


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

Re-INVITE – Answer in ACK – Decline Video Support

Troubleshooting Tools


SIP Troubleshooting Tools

Unified CM / SME Tools:

- Real Time Monitoring Tool / Session Trace

- TranslatorX

Wireshark


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

Can only be used on calls for which traces still exist on the server (Unified CM 9.0 will allow viewing traces that have been archived off-server)

Session Trace Features




RTMT Session Trace Tool Call Flow Diagram



Hover to get information about message

Click “See SIP message” to see actual SIP message


RTMT Session Trace Tool SIP Message Details


TranslatorX Tool

Parses through Unified CM CCM/SDI Trace Files

Decodes SIP, SCCP, H.323, MGCP, 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://www.employees.org/~pgiralt/TranslatorX

NOTE: Do not call TAC for support on TranslatorX

Features

http://www.employees.org/~pgiralt/TranslatorX


TranslatorX Tool


TranslatorX Tool Call List Window


TranslatorX Tool Call List Filtering

Select Each Call and click

“Generate Filter” button


TranslatorX Tool CDR View


TranslatorX Tool CDR View


TranslatorX Tool Message Filters


TranslatorX Tool


TranslatorX Tool


Wireshark

Open Source network packet capture and analysis tool

Available at http://www.wireshark.org

Available for Windows, Mac OS X, and UNIX/Linux

Provides VoIP Call and SIP analysis

http://www.wireshark.org


Wireshark


Wireshark

VoIP Call Analysis


Wireshark

VoIP Call Ladder Diagram


Wireshark

Both Unified CM and IOS provide a mechanism to gather a packet capture

Will be covered later

How to gather a trace?

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

Configured from Cisco Unified Serviceability > Trace > Configuration or by using AnalysisManager

Unified CM 9.0 will combine SDI and SDL traces into the SDL traces, so be on the lookout.



Select the

Server

Select the Service on

Which Trace Needs to

Be Enabled

Select Service

Group



1. Press

Set Default

2. Set to Detailed

Updates All

Servers in This

Cluster with

These Settings



Enable SIP Stack Trace is NOT needed to see SIP Messages.

Do not enable SIP Stack Trace unless directed by TAC



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


Trace Collection

RTMT Analysis Manager

RTMT Remote Browse

RTMT Collect Files

RTMT Query Wizard

OS CLI (file get or file tail)

Various Ways to Collect Trace Files


Gathering a Packet Capture from Unified CM

Use the Platform CLI command „utils network capture‟

admin:utils network capture ?

Syntax:

utils network capture [options]

options optional page , numeric, file fname, count num, size bytes, src addr,

dest addr, port num, host protocol addr

admin:utils network capture file capturefile count 100000 size ALL host ip 10.1.1.1

Executing command with options:

size=ALL count=100000 interface=eth0

src= dest= port=

ip=10.1.1.1

admin:file list activelog platform/cli

capturefile.cap

dir count = 0, file count = 1

admin:file get activelog platform/cli/capturefile.cap

Please wait while the system is gathering files info ...done.

Sub-directories were not traversed.

Number of files affected: 1

Total size in Bytes: 24

Total size in Kbytes: 0.0234375

Would you like to proceed [y/n]? y

Cisco Unified Border Element (CUBE) Tracing Configuration


CUBE Debugging

CUBE / IOS Tools:

- IOS debugs

- IOS show commands

- Per-call trace

- Packet export


CUBE Debugging

When debugging in IOS, configure logging buffered to a fairly large value (based on available memory)

Disable logging to the console with command „no logging console‟

Enable timestamps for debugs

Make sure router has NTP enabled

service timestamps debug datetime msec localtime

service timestamps log datetime msec localtime

logging buffered 10000000

no logging console

clock timezone EST -5 0

clock summer-time EDT recurring

ntp server 10.14.1.1


CUBE Debugging

Various SIP debugs available:

CUBE#debug ccsip ?

all Enable all SIP debugging traces

calls Enable CCSIP SPI calls debugging trace

dhcp Enable SIP-DHCP debugging trace

error Enable SIP error debugging trace

events Enable SIP events debugging trace

function Enable SIP function debugging trace

info Enable SIP info debugging trace

media Enable SIP media debugging trace

messages Enable CCSIP SPI messages debugging trace

preauth Enable SIP preauth debugging traces

states Enable CCSIP SPI states debugging trace

translate Enable SIP translation debugging trace

transport Enable SIP transport debugging traces

verbose Enable verbose mode


CUBE Debugging Sample „debug ccsip messages‟

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

Received:


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





Cisco-Guid: 3720668288-0000065536-0000015564-0996807340


P-Asserted-Identity: "Paul Giralt" <sip:[email protected]>


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)


CUBE show commands

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

0 : 2807 92135710ms.1 (23:55:20.115 EST Mon Jan 16 2012) +1770 pid:1 Answer 89915644 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


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

Reference


CUBE – IP Traffic Capture

IP Traffic Export feature allows export of packets on an interface

Configuration:

Export Packet Data in PCAP format

ip traffic-export profile CUBE_Debug mode capture

bidirectional

incoming access-list 101

outgoing access-list 101

interface GigabitEthernet0/0

ip traffic-export apply CUBE_Debug size 10000000

Usage:

traffic-export interface g0/0 start

traffic-export interface g0/0 stop

traffic-export interface g0/0 copy scp://10.1.1.1/capture.pcap

Case Studies


Case Study 1: Unable to place a call

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

Problem Description



Use RTMT Session Trace

Enter *5551212 into Called Number/URI field

Set time and duration appropriately

Search Finds two calls




Double-click to see message diagram

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



Troubleshoot call on CUBE

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


b82e2c213ca7-45568313


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



Min-SE: 1800



CSeq: 101 INVITE

Expires: 180



Cisco-Guid: 3852191232-0000065536-0000018595-0996807340


P-Asserted-Identity: "Paul Giralt" <sip:[email protected]>

Contact: <sip:[email protected]:5060>

Max-Forwards: 69

Content-Length: 0



Troubleshoot call on CUBE

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


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

Check to see if the number matches a valid dial peer


Case Study 2: Unable to place a call #2

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

Problem Description




Enter *5551212 into Called Number/URI field

Set time and duration appropriately

Search Finds two calls




Trace shows signaling from phone as well as to CUBE

CUBE is responding with a 403 Forbidden



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

Problem Description

voice service voip

no ip address trusted authenticate

allow-connections sip to sip

sip

voice service voip

ip address trusted list

ipv4 172.18.106.0 255.255.255.0

allow-connections sip to sip

sip


Case Study 3: No One Answers the Phone

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

Problem Description



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

Collect Traces



Problem is reproducible, so generate a test call and then collect traces. Select File > Open Folder…

Use TranslatorX



Search for called party number

Use TranslatorX to Analyze Traces



Disable Filters

Select the INVITE

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




03/29/2010 10:36:41.497 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to

172.18.159.231:[5060]:



From: "Test User 1" <sip:[email protected]>;tag=97903bc0-a3de-4a15-ba27-44c81fe3adcd-

45510543


Date: Mon, 29 Mar 2010 14:36:41 GMT



Min-SE: 1800



CSeq: 101 INVITE

Expires: 180





Cisco-Guid: 2081204224-3137452793-0000000466-0996807340




Max-Forwards: 69

Content-Length: 0




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




Select the INVITE

Create New Filter (control/command-N)

Filter by IP Address (control/command – I)

Re-enable Filters







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>


Max-Forwards: 70

Date: Mon, 29 Mar 2010 14:36:33 GMT

CSeq: 101 INVITE

User-Agent: Cisco-CP9951/9.0.1


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-Disposition: session;handling=optional

INVITE from IP Phone w/ SDP



v=0


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=fmtp:116 mode=20



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

INVITE from IP Phone w/ SDP (continued)



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




Date: Mon, 29 Mar 2010 14:36:33 GMT


CSeq: 101 INVITE


Content-Length: 0

Unified CM Sends a 100 Trying



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



CSeq: 101 REFER

Max-Forwards: 70



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

Unified CM Sends a REFER to play Outside Dialtone



<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>

Unified CM Sends a REFER to play Outside Dialtone (continued)



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




CSeq: 101 SUBSCRIBE

Date: Mon, 29 Mar 2010 14:36:33 GMT


Event: kpml; [email protected]; from-

tag=00260bd9669e07147bcb3aac-3cda8f0c

Expires: 7200


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>

Unified CM Sends a SUBSCRIBE for KPML





SIP/2.0 200 OK

Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK151511c5f04bf


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

Phone Sends 200 OK for the REFER and SUBSCRIBE



SIP/2.0 200 OK



To: <sip:[email protected]>;tag=00260bd9669e07177ee0d51d-14f56f89


Date: Mon, 29 Mar 2010 14:36:33 GMT

CSeq: 101 SUBSCRIBE



Expires: 7200

Content-Length: 0





SIP/2.0 200 OK



To: <sip:[email protected]>;tag=00260bd9669e07177ee0d51d-14f56f89


Date: Mon, 29 Mar 2010 14:36:33 GMT

CSeq: 101 SUBSCRIBE



Expires: 7200

Content-Length: 0

Unified CM Sends a SUBSCRIBE for KPML



IP Phone (172.18.159.152)

Unified CM (172.18.159.152)

SIP Gateway (172.18.159.231)

INVITE

100 Trying

REFER

SUBSCRIBE

200 OK (REFER)

200 OK (SUBSCRIBE)





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


Date: Mon, 29 Mar 2010 14:36:33 GMT

CSeq: 1001 NOTIFY

Event: kpml

Subscription-State: active; expires=7200

Max-Forwards: 70


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"/>

User Dials a „1‟





SIP/2.0 200 OK




Date: Mon, 29 Mar 2010 14:36:34 GMT


CSeq: 1001 NOTIFY

Content-Length: 0

Unified CM Replies to NOTIFY With a 200 OK






Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK151536ea86ab0




CSeq: 101 REFER

Max-Forwards: 70



Expires: 0



Require: norefersub



Content-Length: 401

Unified CM Replies Sends a REFER to Disable Outside Dialtone



<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>










CSeq: 101 REFER

Max-Forwards: 70



Expires: 0



Require: norefersub



Content-Length: 401

Unified CM Replies Sends a REFER to Disable Outside Dialtone





SIP/2.0 200 OK



To: <sip:[email protected]>;tag=00260bd9669e07184b08b96b-796ab86f


Date: Mon, 29 Mar 2010 14:36:33 GMT

CSeq: 101 REFER



Content-Length: 0

Phone Replies With 200 OK to REFER



IP Phone (172.18.159.152)

Unified CM (172.18.159.152)

SIP Gateway (172.18.159.231)

INVITE

100 Trying

REFER

SUBSCRIBE

200 OK (REFER)

200 OK (SUBSCRIBE)

NOTIFY

200 OK (NOTIFY)

REFER

200 OK (REFER)



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




Date: Mon, 29 Mar 2010 14:36:34 GMT

CSeq: 1002 NOTIFY

Event: kpml

Subscription-State: active; expires=7195

Max-Forwards: 70


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

Content-Length: 209

Content-Type: application/kpml-response+xml


<?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"/>

User Dials a „8‟



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




Date: Mon, 29 Mar 2010 14:36:34 GMT


CSeq: 1001 NOTIFY

Content-Length: 0

Unified CM Replies to NOTIFY With a 200 OK



User Dials Remaining Digits



IP Phone (172.18.159.152)

Unified CM (172.18.159.152)

SIP Gateway (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



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

Digit Analysis Match



|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




|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

>

}

}




|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=




|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

|AlternateMatches=




03/29/2010 10:36:41.497 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to

172.18.159.231:[5060]:




45510543


Date: Mon, 29 Mar 2010 14:36:41 GMT



Min-SE: 1800



CSeq: 101 INVITE

Expires: 180





Cisco-Guid: 2081204224-3137452793-0000000466-0996807340




Max-Forwards: 69

Content-Length: 0

CUCM Sends an INVITE to the PSTN Gateway



IP Phone (172.18.159.152)

Unified CM (172.18.159.152)

SIP Gateway (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)



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



45510543

To: <sip:[email protected]>;tag=DE1EFF8-0

Date: Mon, 29 Mar 2010 14:37:23 GMT


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

Gateway Replies With a 183 Session Progress W/ SDP





v=0

o=CiscoSystemsSIP-GW-UserAgent 0 7954 IN IP4 172.18.159.231

s=SIP Call

c=IN IP4 172.18.159.231

t=0 0

m=audio 27980 RTP/AVP 0 8 116 18 100 101

c=IN IP4 172.18.159.231




a=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=fmtp:101 0-16

--uniqueBoundary

Content-Type: application/x-q931

Content-Disposition: signal;handling=optional

Content-Length: 11

Gateway Replies With a 183 Session Progress W/ SDP





SIP/2.0 180 Ringing



To: <sip:[email protected];user=phone>;tag=97903bc0-a3de-4a15-ba27-44c81fe3adcd-45510542

Date: Mon, 29 Mar 2010 14:36:33 GMT


CSeq: 101 INVITE




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

Unified CM Sends a 180 Ringing to the IP Phone



IP Phone (172.18.159.152)

Unified CM (172.18.159.152)

SIP Gateway (172.18.159.231)

INVITE

100 Trying

REFER

SUBSCRIBE

200 OK (REFER)

200 OK (SUBSCRIBE)

NOTIFY

200 OK (NOTIFY)

REFER

200 OK (REFER)

NOTIFY

200 OK (NOTIFY)

NOTIFY / 200 OK

Repeats 10 Times

SUBSCRIBE

INVITE

100 Trying

200 OK (SUBSCRIBE)

180 Ringing

183 Session Progress



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

No SIP Signaling for 50 seconds

Phone Keeps Ringing





CANCEL sip:[email protected];user=phone SIP/2.0





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

Phone Sends a CANCEL





SIP/2.0 200 OK




Date: Mon, 29 Mar 2010 14:37:32 GMT


CSeq: 101 CANCEL

Content-Length: 0

Unified CM Sends a 200 OK for the CANCEL



IP Phone (172.18.159.152)

Unified CM (172.18.159.152)

SIP Gateway (172.18.159.231)

NOTIFY

200 OK (NOTIFY)

CANCEL

200 OK (CANCEL)

CANCEL

200 OK (CANCEL)

487 Request Cancelled

487 Request Cancelled

ACK

ACK



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 “SEP Rel1XX Options” *

-Disabled

-Send PRACK for all 1xx Messages

-Send PRACK if 1xx Contains SDP

*Service Parameter in 7.x and earlier. SIP Profile parameter in 8.x and later

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Documents

Troubleshooting SIP with Cisco Unified Communicationsd2zmdbbm9feqrf.cloudfront.net/2012/eur/pdf/BRKUCC-2932.pdf · BRKUCC-2932 Troubleshooting SIP with Cisco Unified Communications