troubleshooting unified communication

BRKEVT-3304Cisco PublicBRKEVT-3304 Cisco Public© 2011 Cisco and/or its affiliates. All rights reserved.

Troubleshooting Unified communication

Manager and Cisco Telepresence Video

Communication Server Integration

BRKEVT -3304

BRKEVT-3304Cisco Public

Troubleshooting Unified communication Manager and Cisco

Telepresence Video Communication Server Integration

Mahesh Anjan

Sr.Software Engineer

Cisco Systems

BRKEVT-3304Cisco Public© 2011 Cisco and/or its affiliates. All rights reserved. Cisco Confidential

3

A brief overview of the deployment will be provided, however, to maximize the material learned, the attendee should be familiar with Cisco telepresence Video communication server and Unified Communication Manager Infrastructure. Attendees should also have a good understanding of SIP and H.323 call flows.

Appendix slides ( Interop call flows, additional case studies)

Session time: 90 minutes

Please ask questions as we go – 3 types of questions:

Questions I’ll answer

Questions I’ll defer to later in the session

Question I don’t know the answer to, outside the scope of our session, or those that consume too much time.

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5

With UC Manager and Cisco Telepresence VCS bridged in enterprise network, the session aims at troubleshooting various interoperability issues that arises between these call controls and take appropriate steps to resolve them

Objective


6

Deployment Overview

UC Manager Integration with Cisco Telepresence VCS

Cisco Telepresence VCS Integration with Unity Connection voicemail

UC Manager Session Layer Bandwidth

UC Manager BFCP Interoperability

Cisco Telepresence Native Interoperability

UC Manager Integration with Cisco Telepresence MCU

UC Manager Integration with ISR G2 Conference Bridge

Cisco Telepresence endpoints Native registration with UC Manager

Troubleshooting Summary

Appendix – Extra case Studies

Agenda


© 2011 Cisco and/or its affiliates. All rights reserved. Cisco Confidential7

Baseline Software version

• 8.6 UC Manager

• X6.1 Cisco Telepresence Video Communication Server

• 4.2 Cisco Telepresence MCU

• 8.6Cisco Unity Connection

• IOS : 15.1.4MISR G2 Video Conference Bridge

• TC 5EX/C Series

• TE 4.0 E Series

• 1.7.4Cisco Telepresence Codec 500/1000/3000

• F 9.0.2 Cisco Telepresence MXP Series

• 3.0.0.1Polycom HDX 4000

• 9.0.6.1Polycom VSX 7000

• 3.4.2Sony PCS-1



Abbreviations

• Unified Communication ManagerUCM

• Video Communication ServerVCS

• Media Conference UnitMCU

• Delayed Offer in UCM ( Invite without SDP)DO

• Early offer in UCM ( Invite with SDP)EO

• Payload Type PT

• Unity Connection Uconn

• Binary Floor Control ProtocolBFCP

• Integrated Service RouterISR

• Transport Independent Application SpecificTIAS

• Application SpecificAS

• Conference TotalCT


9

Deployment Overview


10

Deployment Overview

DO:SIP Trunk

MXP 1700 H323/SIP

MOVi-SIP

E20-SIP

Ex90

H323/SIP

E20-SIPC20-SIP

Ex90-SIP

CUCIlync

CIUS-SIP

MCU 4205

VCS

SP

MCU

SONY-PC1-H323

9971

SONY-PC1-H323

Polycom VSX 7000

SIP Trunk

SIP Registration

H225 Trunk


11

Related Sessions• For more information about UCM – VCS Deployment consider attending:

BRKEVT-2315 Understanding the Cisco Tandberg Infrastructure

Thursday, July 14th @ 8:00AM

• For more information about Deploying IP Video Telephony Networks Consider attending :

BRKEVT-2300 Designing and Deploying IP Video Telephony Networks

Tuesday, July 12th @ 12:30 PM

• For more information about Video Quality troubleshooting Refer to:

BRKEVT- 3934 Troubleshooting Voice and Video Quality in Unified Communication Network

Monday, July 11th @ 8:00AM

• For more Information about UC Manager Serviceability Troubleshooting Refer to:

TECUCC-3000 Cisco Unified Communications Manager Serviceability and Troubleshooting

Monday, July 11th @ 8:00AM


12

Log collection from VCS , VCS endpoints & UC Manager



Diagnosing VCS Issues

How do I access VCS and what are the Common Logs to Collect

• Connect the serial cable from the DATA port on the front of the VCS to the serial port on a PC.

• baud rate : 115200 bits per second

• data bits : 8

• Parity : none

• stop bits : 1

• flow control (hardware and software): none

Serial Console connection

• Root Login (default) : root / TANDBERG

• Telnet (default) : admin / TANDBERG

SSH , Telnet Access

• SSH to the endpoint and enter netlog 2Protocol level

decode

• SSH to the endpoint and enter netlog 3Protocol level

decode,hex dump

• netlog offxcom log network debugxcom log developer.iwf debugconsole onxcom log network warnxcom log developer.iwf warn

SIP – H323 Interworking logs



• SSH using root username and TANDBERG as password , If root access is disabled , SSH using GUI username and password and then configure system tools rootsettings on [password] , this will set root password on the unit.

Enabling Root Access

•root Login (default) : root / TANDBERG

•Telnet (default) : admin / TANDBERGSSH , Telnet Access

• set-a qwertsys

• syslog off

• sys-f

• s-t 0x1300 upr aciptel acipadm h323m uis uic

• com h323m 0 log call rtp

• Quit

Logs from VCS MXP 1700 h323 endpoint

•Login as "admin" via SSH and then enable "log ctx SipPacket debug 9"

SIP logs from Ex/C Series

•Login as "admin" via SSH and then enable “log ctx H323Packet debug 9”

Logs from VCS Ex/C series h323 endpoint

•Login as "admin" via SSH and then enable

• log ctx SipPacket debug 3

• log ctx MC debug 3

• log ctx APPL debug 3

Collecting Logs from E20

•Login as "admin" via SSH and then run the command

• tcpdump -n -s 1500 -w /tmp/log.pcap ip and not port 22

Collecting tcpdumpfrom endpoints

Diagnosing VCS Endpoint Issues

What are the Common Logs to Collect



Diagnosing VCS Endpoint Issues

Collecting logs from E20

• Taking screenshots on E20

• Login using SSH and enter tshell

• mkdir snapshots

• CD /tmp/snapshots

sys-fcom cuilfsm 0 event MenuMagic CreateScreenShot “Current /tmp/snapshots/test.png“ Copy the file from the endpoint using scp root@e20:/tmp/snapshots/test.png

Taking Screenshots on E20

•can be done by logging into the endpoint with ssh root@ip and doing an /etc/init.d/S90osd restart Power cycle via CLI

• Verify IP connectivity (e.g. ping). Get this working first.

• Ensure E20 does not have both HTTP and HTTPS enabledCannot access web page

• The E20 saves logs in the /var/log/eventlog folder, where we have an application, main, video and audio logs. In the case of a reboot or a crash, logs are stored in the /config/logs folder so it may be useful to gather logs from here in the case of bug reporting.

• Log into the system via SSH/Telnet (default username admin, no password)

• The log from the previous session (ie. the session before the system has rebooted) will be /config/logs/log.tar.gz

• In the case of a crash, it could be useful to not only tar up the eventlog folder, but the whole log folder

• The log from the current session is in /var/log, to get it and then scp it do the following:

• $tar zcvf /tmp/current.tar.gz /var/log/eventlog

• $scp /tmp/current.tar.gz user@remotebox:current.tar.gz

Retrieving Logs via SSH/Telnet



UC Manager Log Collection

Collect UCM Logs from

RTMT , Go to System

Trace and Log Central

Collect Files

From UCM Serviceability

Go to Trace Configuration

Ensure SIP Call Processing trace

and SIP Stack Traces are checked

1

2

For more Info on UC Manager log Collection and Troubleshooting , Refer

TECUCC-3000 - Cisco Unified Communications Manager Serviceability and Troubleshooting



UC Manager Log Collection

Wireshark Capture

via UCM CLI

2

3

Cli to Capture

pcap trace


18

UC Manager Integration with Cisco Telepresence VCS


Case Study 1 : 9971 gets fast busy and the call does not get connected


• Topology

9971 UCM SIP Trunk E20

• Logs to Collect

• UCM SDI / SDL Logs

• Sniffer trace from UCM

• Cause Codes from UCM trace

• 503 Service unavailable or might encounter

• 408 Request timeout

SIP Trunk( DO / EO)

VCSUCM

9971

E20

503 Service Unavailable

408 Request time out



• SIP Trunk Failure

What’s the Root Cause ?


Debugging SIP Trunk Failure


If OPTIONS ping is enabled under SIP Profile of the SIP Trunk that connects to VCS.

• UCM traces will log :SIPTrunkOOS alarm in sdi traces

Snapshot of sdi trace : %UC_CALLMANAGER-3-SIPTrunkOOS:

%[DeviceName=options_ping_to_tbd3][UnavailableRemotePeersWithReasonCode=10.4.21.75,

5060, remote=503][AppID=Cisco CallManager][ClusterID=tbd7-pub2-7825I][NodeID=tbd7-pub2-

7825I]: All remote peers are out of service and unable to handle calls for this SIP trunk

SIP Profile

Config

Ensure Zone configuration in VCS shows UCM peer status to be Active and

Zone Profile is set to Cisco Unified Communication Manager

Link to

UCM shows

green



Debugging SIP Trunk Failure Trunk reset/restarted or if there a network issue while the call is active

Failure Code : UCM sends BYE with cause code 41 UCM IP

Cause

Code



Debugging SIP Trunk Failure

SIP OPTIONS

PING from VCS

OPTIONS PING



Case Study 2 : Call from H323 VC endpoints results in One way Video & Two way Audio

• Topology :

• Sony PCS GK UCM VCS E20

• Logs to Collect

• UCM SDI/SDL Logs

• Sniffer Trace from UCM switch port

D0: SIP Trunk

UCMVCS

E20

Display : No Incoming VideoSONY-PCS1

No Video

Audio



Debugging One way Video

INVITE from UCM to VCS

200ok from VCS to UCM: Payload 97 for H264 MIME type

PT:97



Debugging One way Video

PT:105

UCM IP

UCM ACK’s back to VCS with 105 Payload for H264 MIME Type



• Dynamic Payload Negotiation Issue




Transition H323 devices behind GK to VCS

VCS do H323 – SIP Interworking

Eliminate RAS Aggregator trunk , UCM and IOS gatekeeper config

VCS will handle H323 registration and CAC

Solution Proposed

D0: SIP Trunk

VCS

E20

Registered

to VCS

UCM

SONY-PCS-1 SONY-PCS-1

Topology : Sony PCS VCS SIP Trunk UCM 99719971



SONY PCS Calls 9971 behind UCM --- >INVITE from VCS to UCM

200OK from UCM to VCS carries PT 97

Successful Call : Two Way video Established

UCM IP

PT: 97



Successful Call : Two Way video Established

VCS ACK’s back to UCM with payload 97 and two way video is established

PT: 97

VCS IP


31

Cisco Telepresence VCS Integration with Unity Connection voicemail



Case Study 3 : Unable to leave voicemail to E20 registered to VCS

DO: SIP Trunk

Call Fwd All

• Topology

• 9971UCMSIP Trunk VCS E20 (Cfwdall)

• E20SIP TrunkUCM SIP Trunk Unity Connection

• Symptom

• Caller gets redirected to Unity connection welcome prompt

• Logs to Collect

• UCMSDI/SDL and Sniffer traces

E20

9971

VCS

UCM

UConn

28100: Pilot Number



SIP Exchange between UCM and VCS INVITE from VCS to UCM

Request-Line: INVITE sip:[email protected]/2.0

Invite from VCS

without Diversion

header

Call Forward Scenario

in UCM case

Diversion Header

Uconn Pilot Number

[email protected]



SIP Exchange between UCM and Unity connection

• Invite from UCM towards unity connection

Request-Line: INVITE sip:[email protected]:5060 SIP/2.0

VCS Case :

Invite from UCM to

Unity Connection

UCM Case :

Invite from UCM to

Unity connection

Subscriber

DN



Cisco Telepresence VCS does not support diversion header




Workaround : Direct SIP Trunk to UC from VCS

DO: SIP Trunk

Create Zone, Search Rule from VCS to Unity connection

Ensure SIP Connection to Unity connection shows active on VCS

51991

Pilot Number : 59000

9971

UCM

VCS

E20

UConnIP

Unity Connection



Request-Line: INVITE sip:[email protected] SIP/2.0 from VCS to Unity connection

Request URI and TO header are different and Unity Connection Considers To header of user portion to direct the caller to 51991 Subscriber mail box

Working logs

VCS IP

Fwd DN

Pilot DN

[email protected]



INVITE from VCS to UCM

Request-Line: INVITE sip:[email protected] SIP/2.0

What really happened in UCM failure case ?

Request URI and TO Header across SIP Trunk

VCS IP

Fwd DN

[email protected]



Request sent out from UCM towards Unity connection

Request-Line: INVITE sip:[email protected]:5060 SIP/2.0

To Header and Request URI Across SIP Trunk

UCM to UC

UC Pilot not

Fwd DN

UCM does not preserve To header and Request URI across SIP Trunk

UC Pilot

[email protected]:5060



• Normalization Script can be Implemented at the UC Manager Trunk level to modify To Header and Request URI

• Diversion header support in VCS is in the roadmap

Solution for UCM Failure Case


41

UC Manager Session Layer Bandwidth



What are the Common Bandwidth Attributes ?

• The Transport Independent Application Specific Maximum (TIAS) bandwidth modifier has an integer bit-rate value in bits per second. The bandwidth value is the maximum needed by the application (SDP session level) or media stream (SDP media level) without counting IP or other transport layers like TCP or UDP

b=TIAS:<bandwidth-value>

• The Application Specific maximum bandwidth is indicated by the modifier "AS“. The interpretation of this attribute is dependent on the application's notion of maximum bandwidth.

b=AS <application Specific>

• Conference Total gives a maximum bandwidth that a conference session will use. Its purpose is to decide if this session can co-exist with any other sessions.

b=CT <conference total>



A Quick Glance

BW Attributes Pre 8.6 8.6 Impact

b= TIAS (b/s)Used as session and media level .

Does not include transport overhead

Processed at

session level and

pass through at media

level

Supported Video Quality dropped

to as low as 300k and

13 fps

b= AS (Kb/s) Ignored processing

b=AS

Supported Some endpoints

recognized only AS as

BW modifier that

resulted in over or

under subscription of

UC Manager region

bandwidth

b= CT (Kb/s) Ignored processing

b=CT

Started

processing

b=CT

Some endpoints

recognized only CT

as BW modifier

modifier that resulted

in over or under

subscription of UC

Manager region

bandwidth



Case Study 4 : Session layer Bandwidth Behavior with UCM 8.5 and 8.6

DO: SIP Trunk

UCM 8.5

• Topology

• E20 UCM SIP Trunk UCM 9971

• E20 calls 9971

• Logs to Collect

• UCM SDI / SDL and Sniffer trace

9971

E20UCM 8.5

9971DO: SIP Trunk

UCM 8.6 E20

UCM 8.6

1

2

9971

9971



UCM 8.5 : Region Bandwidth set to 384 Kbps

Initial INVITE from E20 to UCM Initial Offer from

E20 to UCM set to

1152

Offer goes in form

of AS




200ok from UCM to E20

No AS

parameter in

the answer

No TIAS for

Audio m-line

384000 – 64000 = 320000




E20 Screenshot

E20 channel rate falls

to very low bit rate

degrading video qualityWill experience choppy or black video on E20



Comparing with UCM 8.6 traces : Region Bandwidth set to 384 kbps

Initial INVITE from E20 to UCM

Default call

set to 1152




200ok from UCM to E20

UCM Indicates

Region

Bandwidth

TIAS: 64000

for G711

384000 – 64000 = 320000 for Video




E20 Screenshot

E20 Adheres

to UCM

bandwidth

Channel rate

does not to a

very low BW

E20 is offered back with AS and E20 adheres to UCM bandwidth management



Session Layer Bandwidth config Snapshot

In the outbound offer sent by UCM, which includes the Early

Offer or re-invite Offer, UCM will send session level bandwidth

modifier based on SIP Profile configuration, and this

configuration can be applied to both SIP line side device and

SIP trunk device.

SIP Profile

config

TIAS and AS

default



UC Manager Offer / Answer Behavior

Caller’s offer: b=AS:384

Callee’s offer: b=CT:512 Region bw = 512

Answer to caller: b=AS:512 Answer to callee: b=CT:384

Caller’s offer:b=AS:384,b=CT:384

Callee’s offer: b=CT:512

Region bw = 512

Answer to caller: b=AS:512

Answer to callee: b=CT:384

Note: If AS and CT were present, CT will be ignored.

Caller’s offer: b=AS:384, b=CT: 384, b=TIAS:384

Callee’s offer: b=TIAS:512

Region bw = 512

Answer to caller: b=TIAS:512

Answer to callee: b=TIAS:384

Note: If TIAS, AS and CT were

present, AS and CT are ignored

Caller’s offer: b=AS:740

Callee’s offer: b=CT:384

Region bw = 512

Answer to caller: b=AS:384

Answer to callee: b=CT:512


53

UC Manager BFCP Interoperability



Understanding BFCP Interop

UCM Cluster UCM Cluster

SIP Trunk EO / DO

SIPSIP

Audio

Video (main)

Video (Presentation)

BFCP

BFCP is supported on Early Offer (EO) and

Delayed Offer (DO) SIP trunks

CUCM never terminates the BFCP protocolUCM never terminates the BFCP protocol

UDP / BFCP Supported : Preferred Industry Standard

TCP /BFCP : Unsupported : Legacy Deployments



Case Study 5 : Unable to View Presentation on C20

TopologyEx90 UCM SIP Trunk VCS SIP C20

Ex90 starts presentation

Logs to Collectnetlog 2 , netlog 3 from VCS

UCM Sdi and SDK Trace

Sniffer traces from UCM

Ex90C-20

UCM

VCSDO SIP Trunk



Debugging Sniffer trace

Ex90 Negotiates

UDP/BFCP

Initial Invite from EX90 to UCM



Debugging Sniffer trace

Invite from UCM to VCS

No UDP/BFCP app

line in SDP




BFCP Disabled

on SIP Profile

Allow Presentation Sharing using BFCP is disabled

Other reason Could be : Flow Control Not negotiating

Server and Control exhibiting same behavior

BFCP capabilities on a

SIP Trunk or SIP Line

can be turned on or

off via SIP Profile

Configuration




BFCP Disabled

on SIP Profile

SIP Profile with

BFCP –disabled

is configured

Allow Presentation Sharing using BFCP is disabled

Other reason Could be : Flow Control Not negotiating

Server and Control exhibiting same behavior

BFCP capabilities on a SIP Trunk or

SIP Line can be turned on or off via

SIP Profile Configuration



VCS Default Communication Manger Profile sets SIP UDP/BFCP filter to “ON”

Is there any Config on VCS Side to enable BFCP ?



when it is “ON” , VCS does TCP/BFCP Which is not supported with UC Manager and not the industry preferred choice





Create a CUSTOM Profile and set the SIP UDP/BFCP filter mode to “OFF”



Working Config

SIP Profile under

SIP Trunk is enabled

with BFCP

After enabling “Allow Presentation Sharing using BFCP” in SIP profile



Working Config

UCM Negotiates

UDP/BFCP App

line

After enabling “Allow Presentation Sharing using BFCP” in SIP profile



BFCP and Security Enabling / Disabling

BFCP Security in UCM

When this parameter is set to True (default), calls utilizing a non-secured BFCP application stream will be treated as secure, provided that the remaining media streams in the call are secured

Negotiated SDP

Audio: Secure

Main Video: Secure

Presentation Video: Secure

BFCP App: Non-Secure

Audio: Secure

Main Video: Secure

Presentation Video: Non Secure


Audio: Secure

Main Video: Secure

Presentation Video: Secure


Service Parameter

True

True

False

Security Status of Call

Secure

Non-Secure

Non-Secure



BFCP Device and endpoint Support BFCP Transmit and Receive BFCP Receive only BFCP Transmit Only

SIP Line Devices

SIP trunk (DO & EO )

EXSeries -SIP ( Registered to UCM /VCS )

E20-SIP ( Registered to UCM / VCS )

MOVi ( Registered to VCS )

MXP 1700 – SIP ( Registered to VCS )

CSeries- SIP ( Registered to VCS / UCM )

UC Endpoints (99xx , CUPC)

Gateways ( SIP , MGCP , H323, SCCP )

CiscoTelepresence MCU

Cisco Telepresence Server

Cisco Telepresence Codecs 1000/3000/500

MXE

UCM VCS

H.239 to BFCP

BFCP to BFCP

Use VCS as

Proxy



Case Study 6 : Getting 481 Call transaction does not exist from VCS

•Topology:

E20 (DN: 51991) VCS SIP Trunk UCM GK Polycom(H323)

•Logs to Collect :UCM SDI/SDL Logs

Sniffer trace

Net log 2 from VCS

D0: SIP Trunk

Polycom-VSX-7000

E20

VCS

UCM

481 SIP Message



Debugging 481 Call Transaction

• UCM sends UPDATE without SDP to provide the called/connected party name and number information during the call setup as well as after call is answered. UCM uses PAI & RPID headers to update this information.

• SIP UPDATE :

P-Asserted-Identity: "Polycom-VSX7000-H323-26002" <sip:[email protected]>

Remote-Party-ID: "Polycom-VSX7000-H323-26002" <sip:[email protected]>;party=calling;screen=yes;privacy=off

180 Ringing had (no display name) –

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

Remote-Party-ID: <sip:[email protected]>;party=called;screen=yes;privacy=off

Remote party



2010-09-03 11:19:31,855 Module="network.sip" Level="INFO":

Dst-ip="10.4.21.11" Dst-port="5060"

Detail="Sending Response Code=481, Method=UPDATE, To=sip:[email protected],

[email protected]"

2010-09-03 11:19:31,855 Module="network.sip" Level="DEBUG":

Dst-ip="10.4.21.11" Dst-port="5060"

SIPMSG:

|SIP/2.0 481 Call/Transaction Does Not Exist

Via: SIP/2.0/TCP

10.4.21.11:5060;branch=z9hG4bK36b215d882ef;received=10.4.21.11

Call-ID: [email protected]

CSeq: 101 UPDATE

From:

<sip:[email protected]>;tag=109550~78e066fc-a56b-45ac-83ce-0a62eb5c06b2-8124

2816

To: <sip:[email protected]>;tag=bae9771a1d8e8596

Server: TANDBERG/4099 (X5.2)

Debugging 481 message

Response from VCS to

UCM



• In response to 481 , UCM responds back with 487 with cause code 127

2010-09-03 11:19:31,875 Module="network.sip" Level="INFO":

Src-ip="10.4.21.11" Src-port="5060"

Detail="Receive Response Code=487, Method=INVITE, To=sip:[email protected],

[email protected]"

2010-09-03 11:19:31,875 Module="network.sip" Level="DEBUG":

Src-ip="10.4.21.11" Src-port="5060"

SIPMSG:

|SIP/2.0 487 Request Cancelled

Via: SIP/2.0/TCP

10.4.21.75:5060;egress-zone=ABICUCMNeighbor;branch=z9hG4bKbb7ef2e619cb05

3c53d5819b5a80736e78322.a5b35826b378cf4a7d515b197978bf9d;proxy-call-id=a

6d049c4-b76e-11df-ab04-0010f31abd40;rport,SIP/2.0/TCP

10.4.22.107:5060;branch=z9hG4bK41018499480896c339bfc702fc974502.1;receiv

ed=10.4.22.107;rport=49665;ingress-zone=localregistration

Call-ID: [email protected]

CSeq: 100 INVITE

From: <sip:[email protected]>;tag=bae9771a1d8e8596

To:

<sip:[email protected]>;tag=109550~78e066fc-a56b-45ac-83ce-0a62eb5c06b2-8124

2816

Reason: Q.850 ;cause=127

Content-Length: 0

Debugging 481 Call transaction does not exist

Response from UCM

to VCS



VCS does not support the UPDATE before answer, or PRACK (Provisional response Acknowledgment)




Configure Custom Profile in VCS version < 6.0 version to strip SIP UPDATE message

Solutions

SIP UPDATE

being stripped off



Solutions

VCS 6.1 Version ignores or suppresses SIP UPDATE messages

VCS will start supporting SIP UPDATE header (in roadmap)


74

Cisco Telepresence Native Interop



What is Native Interop ?

CTS - 1000

MXE 5600

UCM

CIUS

EX90E209971

SIP

Tru

nk

EX90

CTS CUCM 8.5 CUCM 8.0

1.7.4 Yes No for

interop calls

CTS now supports

unrestricted media and

handles variable frame

rate and timestamps.



Sysop Logs : Is the call interop or non-interop ?

2011-04-25 03:50:27: INFO Connected to Senthil-12006 at 12006

2011-04-25 03:50:28: INFO Negotiated Media Configuration: LTRP: Enabled GDR: Enabled High Profile: Disabled Cabac: Enabled

2011-04-25 03:50:28: INFO Endpoint used protocol MUX version 7 for this call

2011-04-25 03:50:28: INFO The presentation has been negotiated to 30 FPS

2011-04-25 03:52:16: INFO Local user dialing 81123136

2011-04-25 03:52:19: INFO Connected to 81123136

2011-04-25 03:50:28: INFO CTS is in an interop call, and is not using TIP or MUX.

Non-Native

Interop

Native Interop

Legacy added for indicating

the interopstatistics

Web interface changes for statistics



How to identify resolution from CTS Logs ?

SSH to CTS codec

login root/ c1sc0cts3

To view resolution of active calls:

cmastat | grep state

CTS:>cmastat | grep state

TBA source pid 2529 channel 0 state off inputs 0x200002 format 0x01000007 dspCmds 206

TBV source pid 2984 device 0 state off imageSize 0x0

TBV sink pid 2526 device 1 state on imageSize 352x288 videoClock 26999768.000000TBV source pid 2985 device 2 state off imageSize 0x0

TBV source pid 2984 device 4 state on imageSize 352x240

state on is the

image resolution

that CTS is

sending

•Sink means receivers

•Source means encoders



Native Interop Summary ( Call from CTS to)

Endpoint Media Signaling Presentation Registration

Ex-series 1080p / 30 SIP (pt 2 pt)

TIP (MultiPt)

BFCP (pt2pt)

TIP (MultiPt)

UCM 8.6

VCS

C-series 720p / 30 SIP (pt 2 pt)

TIP (MultiPt)

BFCP (pt2pt)

TIP (MultiPt)

UCM 8.6

VCS

E-20 w360p sent*

w448p received(*E20 can recv up to448p)

SIP BFCP

receive

CUCM 8.5

VCS

MOVi 720p / 30 SIP BFCP VCS

MXP w360p sent*

w448p received*(MXP can send recv720p)

SIP(H.323 on VCS)

Switched VCS

Cius 720p / 30 SIP NA UCM

CSF clients 720p / 30

or w360p

SIP NA UCM

9951, 9971 w360p sent*

w360p received(*99xx can recv up to 448p)

SIP NA UCM

8941, 8945 CIF, initially*(*w360p send/recv post-8.6, 89xx

can be sent up to w448p)

SIP(SCCP or SIP on

UCM)

NA UCM

3rd party (Polycom) 720p / 30 SIP(H.323 on VCS)

BFCP(H.239 on VCS)

VCS



How does CTS uses the max-fs to choose the encoding resolution

Max-fs Resolution

96 – 919 352x288

920 – 3599 640x360

3600 – 8159 1280x720

> 8159 1920x1080

How bitrates affect the resolution

Resolution Bitrate Max-fs

1920x1080 > 3M > 8159

1280x720 > 936Kbps 3600 – 8159

640x360 > 500Kbps 920 - 3599

352x288 >100Kbps 96 - 919

Resolution , Video Quality and max-br mapping



Quality (per Display) max-br (kbps)

Highest detail, Best Motion : 1080p 4000

Highest Detail, Better Motion:1080p 3500

Highest Detail, Good Motion:1080p 3000

High Detail, Best Motion : 720p 2250

High Detail, Better Motion: 720p 1500

High Detail, Good Motion : 720p 1000

High Detail, Limited Motion : 720p (Lite) 936

Limited Bandwidth Mode : 360p (For TIP Call) : 30fps 475

Cisco telepresence Video Quality & max-br mapping

Codec

Config

in UCM

Max-

res:720

p


81

UC Manager Integration with Cisco Telepresence MCU



MCU Conference Overview

E20-SIPC20-SIP

Ex90-SIP

CUCILync

CIUS-SIP

MXP 1700 SIP

MCU

Intercluster SIP Routing

UCMUCM

CUPC

E20

C20



• MCU is added to UCM via Media Resources Conference Bridge as Cisco Telepresence MCU under

How do we add MCU to UCM ?



Registration Status

MCU

Registered

Http

credentials of

MCU



• MCU resource is added to MRG and then MRGL




• MCU contained MRGL is configured as part of phone MRGL


Phone MRGL



MCU Side Config

Media Port Reservation was changed to "Enabled“



MCU Side Config

Incoming calls to unknown conferences or auto attendants was changed to "Disconnect Caller“

Time to wait when setting up ad hoc conference PIN was changed to "never configure PIN"



How to Debug MCU Registration Issues

Verify that the MCU address is correct

Verify that the SIP port incoming, and Outgoing match what is configured on the MCU

Verify that the MCU is on line.

Verify that the MCU is configured for SIP.

Check the UCM traces for the OPTIONS ping to the MCU

MCU

Stattus

Ensure MCU shows registered in RTMT under UCM Cluster Media resources



Case Study 7 : MCU Adhoc bridge in UCM allocates only Video ports not Audio

Adhoc conference

• Topology

9971 UCM 9971

|__Conference :conference Button 7945

• Logs to Collect

MCU Logs

Non Video

phone

9971

9971

7945

MCU



Debugging Port Usage on Codian

Non Video

phone



Debugging MCU Logs Initial DO Invite from

UCM to MCU

MCU offers both

Audio and Video

Caps for Initial late

media

Video port Allocated



UCM will always include video in the ACK to the Cisco Telepresence MCU because the UCM treats the MCU as pure video bridge




• While the conference is Active, modify conference audio port limit on configuration tab

Workaround to use only audio ports for audio-only

capable endpoint

Initial Call to

MCU

video port

Allocated

Audio-only

capable endpoint

reserve only

audio port

Audio port

getting used



UC Manager will request Video or Audio specific port based on the phone capability ( is in the roadmap)

Alternate Solution


96

UC Manager and VCS Multiway Conference



Case Study 8 : Multiway URI Conference Fails when UC endpoints are Conferenced

SIP : Multiway Conference

C20 - Multiway Conference

• Topology

C20 UCM 9971

C20 Multiway Conference 9971

C20 Multiway Conference VCS Codian

• Logs to Collect

UCM SDI /SDL , Sniffer Trace

VCS

Codian MCU

9971

9971UCM



VCS and MCU Side config

Conference alias

MCU Conference

alias config



UCM Side Routing config

REFER from VCS

to UCM when

multiway is

invokedRHS matches

TLD and SIP

Route pattern

UCM & VCS Domain : abi.com



How UCM interprets SIP URIs

Is RHS the

IP address

of a cluster

member?

no

Analyze LHS

Does RHS

match

CFQDN?

yes yes

no Does RHS

match

OTLD?

yes

Does LHS

find a

match?

yes

Offer call

Match RHS

against

SIP Route

Pattern

no

no

Route or

block

Route or

block

user @ mydomain.org

left-hand-side (LHS) right-hand-side (RHS)



VCS sends 302 with multiway factory number to join the bridge

What happens when C20 selects “join Calls”

VCS to UCM : 302

REFER TO

Header



UCM does DNS Query of host part in the sip contact header i.e abi.com

UCM Fails to do digit analysis to the redirected contacts and the call does not get routed correctly

Debugging multiway

Redirect by Stack

DNS IP configured

under UCM platform

DNS resolution fails

abi.com does not get

resolved to IP address

Search in DNS



One of the Root Cause

Redirect by Application under SIP Profile of the SIP trunk is Unchecked

Ensure Redirect by

Application is checked



With Redirect by Application checked

Solution via redirect by application




VCS sends 302 with multiway factory number to join the bridge

Solution via redirect by application

VCS to UCM : 302




UCM Does digit analysis to the redirected contacts and makes sure that the call get routed correctly

UCM matches SIP route pattern and sends INVITE to VCS with URI set to “Refer to” contact header

Working logs NO DNS Query

UCM sends INVITE

to LHS of REFER

header

VCS IP



Active Multiway Conference

Active Registration

All 3 participants are in Multiway conference

MCU Screenshot


108

UC Manager Integration with ISR Conference Bridge



• UCM call control

• Supports H264 AVC: Baseline Profile, level 1.0 / 1.3 /2.0/ 2.1/ 3.0 / 3.1

• Resolution: QCIF; CIF; w360p; VGA; w448p; 4CIF; 720p(Switchingonly)

• Audio Codec : G.711alaw, G.711ulaw, G.729ab, G.729a, G.729b, G.729, G.722r64, and ILBC

Configuring Conference bridge in ISR G2HQ/Campus

UCM

Video

conferencing

Branch

WAN



UCM Config : Heterogeneous & Homogenous Mode

Adding New

conference

Bridge

Added to UCM via Media Resource Conference Bridge



Registration Status with UCM

Conference

Bridge

registered



Mode 1 : Homogenous mode

dspfarm profile 1 conference video homogeneous

router(config)#

codec h264 w360p

maximum conference-participants 16

maximum sessions 10

associate application SCCP

no shut

router(config-dspfarm-profile)#

sccpccm1.2.3.4 identifier 1 version 7.0

sccp

sccpccm group 1

router(config)#

Same Phone type Being Used

Ex: two 9971

debug dsp-resource-manager flex video

debug vxcmsp error



Mode 2: Heterogeneous mode

dspfarm profile 2 conference video heterogeneous

router(config)#

codec h264 w448p

codec h264 vga

codec h264 cif

maximum conference-participants 16 video-cap-class 3

maximum sessions 2

associate application SCCP

no shut

router(config-dspfarm-profile)#

sccp ccm 1.2.3.4 identifier 2 version 7.0

sccp

sccp ccm group 2

router(config)#

Different Phone models used


114

Cisco Telepresence endpoints Native registration with UC Manager



Cisco Telepresence EX/C-Series Native Registration with UC Manager 8.6

EX90

UCM

SIP

• Topology

Ex90 SIP UCM

• C20 SIP UCM

• Logs to Collect :

UCM SDI / SDL Logs

UCM Sniffer Trace

C20

UCM

SIP1

2



Debugging EX/C-Series 90 Registered Native to UCM

SIP

TFTP

UCMURI

From EX/C-Series Web Interface : http://<ex/c-series_ip>



Debugging EX/C - Series

EX/C-Series shows Not Registered after UCM upgrade

• On the touchpad, go to More > Settings > Admin Settings > SIP Settings and change the Proxy address from the hostname to the IP address of the UCM.

EX/C-Series Factory Reset

• xcommand SystemUnit FactoryReset Confirm: Yes

Troubleshooting Restart/ApplyConfig

• If restart or apply config doesn't work from the CCMAdmin pages check the EX90 config and make sure it is in UCM mode: Advanced Configuration->Provisioning->Mode: UCM. Also verify the Provisioning->ExternalManager is set to the TFTP server of your UCM.


118

Troubleshooting Summary


…So far we leaned how to troubleshoot:

UCM – VCS SIP Trunk failure

VCS Interop with Unity connection voicemail

Session Layer Bandwidth Attributes

BFCP interop with UCM and VCS

Cisco Telepresence Native Interop with UC and

VCS endpoints

MCU Video port allocation and Multiway with

UC endpoints

ISR G2 Conference bridge setup

Cisco telepresence Native registration issues



• Check out VCS – UCM Deployment Document

http://www.cisco.com/en/US/docs/telepresence/infrastructure/vcs/config_guide/Cisco_VCS_Cisco_Unified_Communications_Manager_Deployment_Guide_CUCM_6-1_7_8_and_X6.pdf

• Multiway Configuration

http://www.cisco.com/en/US/docs/telepresence/infrastructure/articles/cisco_telepresence_mcu_configure_multiway_kb_166.shtml

• UCM and MCU configuration

http://www.cisco.com/en/US/docs/telepresence/infrastructure/articles/mcu_send_receive_sip_calls_cucm_kb_346.shtml

• Cisco unity connection administration

http://www.cisco.com/en/US/docs/voice_ip_comm/connection/8x/administration/guide/8xcucsagx.html

• ISR G2 Conference configuration

http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps4952/qa_c67-649850.html

Additional Resources For Your Reference









http://www.cisco.com/en/US/docs/voice_ip_comm/connection/8x/administration/guide/8xcucsagx.html





121

APPENDIX


122

Endpoint feature and Interop Summary



Endpoint features and Interop Summary

Endpoints Call Control SharingSecure

Video

Max TX:RX

ResolutionTIP

Phone + CUVA UCM CIF:wCIF

9951/71 UCM future future VGA:w448p not in 8.6

8941/45 UCM future futureVGA:w448p

(w360p future)not in 8.6

Cius UCM webex future 720p not in 8.6

CUCILync, CUPC,

CUCIConnectUCM webex future 720p not in 8.6

CTS UCMTIP

BFCP* 1080p

E20

(VCS/UCM)Either BFCP receive / future w448p not in 8.6

EX/C Series

(VCS/UCM)Either

BFCP,

TIP in multi-pt /future1080p EX

720p C

MXP VCS H.263 switched 720p / w448p

Movi VCS BFCP 720p not in 8.6


124

UCM and Cisco Telepresence VCS Video / Audio Security



• Topology

CTS(Encrypted) UCM SIP Trunk (sRTP/TLS) VCS Ex90 (Encrypted)

• Logs to collect

UCM SDI / SDL & netlog2 from VCS

Configuring Security between UCM and VCS

D0: SIP Trunk : sRTP/TLS

VCS

CTS - 1000EX90

UCM



Configuring Security on VCS Side

SIP Mode Settings

Active on 5061 Port



Configuring Security on VCS Side

Upload Private and Server Certificates

Register Secure

endpoint with VCS



UCM Side Config

1

2

3Should match

VCS Cert x.509



UCM Side Config

4

Port has to

be 5061

5

LUA

Script

Make sure LUA Script that supports best encryption is uploaded to UC Manager

LUA script can be downloaded from : http://www.cisco.com/cisco/software/release.html?mdfid=283423434&flowid=21301&softwareid=282074295&release=8.5&reli

nd=AVAILABLE&rellifecycle=&reltype=latest



Determining Security from UCM TracesUCM Invite

VCS

Response


131

Additional Case Studies



Case Study 9 : Audio Codec Negotiation Failure on MXP 1700

D0: SIP Trunk- G729VCS

MXP 1700E20

Topology:

1700 ( 3rd Party Advanced ) UCM SIP Trunk VCS SIP E20

UCM Phone & Trunk region is set to G729

Logs to collect :

UCM SDI / SDL

UCM

MXP 1700



UC Manager Trace :

000060889 |2010/08/10 13:37:10.395 |100 |SdlSig-I |MXCapabilitiesIncoming

|waitInterfacesCapabilities |MediaExchange(4,100,130,18) |H245Interface(8,100,173,28)

|8,100,17,28.2^10.4.71.11^Port 43211 |[R:N-H:0,N:1,L:0,V:0,Z:0,D:0] Caps[2] =

11(20),12(20)VideoCapCount[3] = 101-0,100-0,107-0extendVideoCapCount[0] =

h245ICTVersion20useOldGWBytesForGSMConversionFcryptoCapCount=0 DTMF Profile (1,2,119,1,F)

000060828 |2010/08/10 13:37:10.378 |100 |SdlSig |SDPOfferInd |waitSDPResponse

|SIPInterface(4,100,62,15) |SIPCdpc(4,100,68,26) |4,100,62,15.1^*^* |[R:N-

H:0,N:1,L:0,V:0,Z:0,D:0] nAudio=1 stackIdx=1 CapCount=4,6(20),4(20),2(20),10(10) port=46784 IP=

ipAddrType=0 ipv4=10.4.22.182 SDPMode=0 SP=F RTP=T SRTP=F idle=F QoS=F nVideo=1 stackIdx=2

CapCount=4,103,101,101,100 port=46786 ipAddrType=0 ipv4=10.4.22.182 SDPMode=0 RTP=T SRTP=F idle=F

nApp=0 numT38Fax=0 MTPAllocated=0 keepAudiomLineForT38=F transID=0 DTMFMethod=2 DTMFConfig=1

RFC2833PT=103 wantDTMF=0 provideOOB=F FCOffer=0 mSipCallState = 1 InactiveSDPReq=F

RSVPLastCollab=F mSIPAccessDevice=F mConfiguredAddrMode=0

000060829

Debugging Payload type from SDI logs

No Codec type 11

MXP Negotiating codec with UCM

UCM Region set to

G729



Determining payload type from UCM traces

Media Payload UCMPayload mapping

G711Alaw64k 2

G711Alaw56k 3

G711Ulaw64k 4

G711Ulaw56k 5

G722_64k 6

G722_56k 7

_G722_48k 8

G7231 9

G729 11

G729AnnexA 12

G.729 Capability

Type

Other codec payload mapping can be found under UCM Documentation

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/service/8_0_2/cdrdef/cdrcodes.html



• Cause code : SDI Snapshot

Outgoing SIP TCP message to 10.4.22.182 on port 49254 index 4

[2122,NET]

SIP/2.0 503 Service Unavailable^M

Via: SIP/2.0/TCP 10.4.22.182:5060;branch=z9hG4bKfb084a45bf5e6347f57d735b4b1c5f21.1;rport^M

From: "21989" <sip:[email protected]>;tag=a3c7d054a3fba9d0^M

To: <sip:[email protected]>;tag=697~78e066fc-a56b-45ac-83ce-0a62eb5c06b2-75907411^M

Date: Tue, 10 Aug 2010 18:37:10 GMT^M

Call-ID: [email protected]^M

CSeq: 101 INVITE^M

Allow-Events: presence^M

Reason: Q.850;cause=47^M

Content-Length: 0^M

UCM Response

Response from UCM

to Endpoint

Cause Code : 503

Service Unavailable



1700 MXP and other MXP products do not support G729, call fails if the region between UCM and the phone or the outgoing trunk is configured to do only 729




Configure Region between MXP (or If you have C-Series) and Outbound trunk to be G711 or G722 (preferred) or any other codec that is listed below

Solution

G.711 G.722 G.722.1 G.728 64 Bit

MPEG4

AACLD

128 Bit

MPEG4

AACLD

Supported Audio Codec

H.261 H.263 H.263+ H.263++

Natural

Video

H.264

Supported

Video Codec



• Topology

E20 SIP VCS H323 GK UCM RT Phone

• Logs to Collect

•UCM SDI/SDL

• Sniffer trace from UCM

Case Study 10: Call fails to connect between 9971 and E20 Via GK Controlled Trunk

SP

9971

UCM VCS

E20



Debugging H225 and H245

In UCM , look for H225 and H245 Negotiation

H245 TCS not coming

from VCS

VCS UCM

Typical H.245 Message

flow

UCM Session timer

times out



H.245 channel has been established already ,UCM has been waiting for the H245 TCS from the VCS side, which never comes back.

What’s the Root Cause



Solution For GK controlled calls from VCS to UCM :

“Wait for Far End H.245 Terminal Capability Set “ should be unchecked on H225 trunk.

(UCM defaults to checked state)

With checked state VCS will not send H245 TCS to UCM and the call will not be established.

Uncheck this box

H.225 Trunk to VCS


Case Study 11 :Video Region Bit Rate (384 kb) < Offer Bandwidth

SIP TBE20

UCM 1 UCM 2

SDP 1

o=tandberg 6 1 IN IP4 10.10.10.33

s=-

c=IN IP4 10.10.10.33

b=AS:1152

m=audio 2326 RTP/AVP 100 101

102 9 18 11 8 0 103

b=TIAS:64000

m=video 2328 RTP/AVP 97 98 99

34 31

b=TIAS:1152000

UCM 1 will take min (b=AS:1152, video region bit

rate) to reserve BW from Location video pool, in this

case, video region is less than Offer bandwidth

value, so 384 kb will be reserved.

SIP 9971

INVITE with SDP

INVITE with SDP

SDP 2

CiscoSystemsCCM-SIP 2000 1 IN

IP4 10.81.55.245

s=-

c=IN IP4 10.10.10.33

b=CT:384


102 9 18 11 8 0 103

b=TIAS:64000

m=video 2328 RTP/AVP 97 98 99

34 31

b=TIAS:320000

UCM 1 sends EO Invite to UCM

2, SIP Profile configured to send

CT

UCM 2 will take min (b=CT:384, video region bit

rate) to reserve BW from Location video pool, in this

case, video region = Offer bandwidth value, so 384

kb will be reserved.

UCM 2 sends delay offer to RT

phone

INVITE w/o SDP

EO SIP

Trunk and

CT Only



Session layer Bandwidth modifier

Case Study 5 : Video Region Bit Rate (384 kb) < Offer Bandwidth

SIP TBE20

UCM 1

200 OK (INVITE)

UCM 2

SDP 3

o=Cisco-SIPUA 27291 0 IN IP4

172.18.203.94

s=SIP Call

t=0 0

m=audio 31490 RTP/AVP 0 8 18

102 9 116 124 101

m=video 31604 RTP/AVP 126 97

c=IN IP4 172.18.203.94

b=TIAS:1000000

RT phone sends Answer with

SDP, no Session Level

Bandwidth, only video line TIAS

SIP 9971

UCM 2 connect media and negotiate codec.

E20 RT direction

will take min (b=TIAS:1000000+negotAudio, video

region bit rate) to adjust Location BW.

RT E20 direction

will take min (b=CT:384, video region bit rate) to

adjust Location BW.

UCM 2 will echo back CT in Answer (200 ok invite),

value = video line TIAS + negotAudio

200 OK (INVITE)

SDP 4

o=Cisco-SIPUA 27291 0 IN IP4

172.18.203.94

s=SIP Call

b=CT:384

t=0 0


102 9 116 124 101

b=TIAS:64000


c=IN IP4 172.18.203.94

b=TIAS:320000

UCM 1 connect media and negotiate codec.

E20 RT direction

will take min (b=CT:384, video region bit rate) to

adjust Location BW.

RT E20 direction

will take min (b=CT:1152, video region bit rate)

to adjust Location BW.



Session layer Bandwidth modifier

Case Study 5 : Video Region Bit Rate (384 kb) < Offer Bandwidth

SIP TBE20

UCM 1 UCM 2

SDP 5

o=CiscoSystemsCCM-SIP 2000 1

IN IP4 10.81.55.241

s=SIP Call

c=IN IP4 172.18.203.75

t=0 0


b=TIAS:64000

m=video 16618 RTP/AVP 98

b=TIAS:320000

ACK with SDP (INVITE)

200 OK w/ SDP (INVITE)

SIP 9971

UCM 2 send Answer (ack invite)

SDP with no Session level

Bandwidth.

Video line TIAS = 384000 –

64000 = 320000

SDP 4

O=CiscoSystemsCCM-SIP 2000 1

IN IP4 10.81.55.245

s=SIP Call

b=AS:384

t=0 0


102 9 116 124 101

b=TIAS:64000


c=IN IP4 172.18.203.94

b=TIAS:320000

UCM 1 send Answer (200 ok

invite) SDP which echo back

Session level Bandwidth AS.


145

Mid call Negotiation Issues



Case Study 12 : One way Video between 8941 and CTS

CTS 1000

• Topology

8941 UCM CTS

• Logs to collect

• UCM via RTMT

• CTS logs from endpoint and look for sipXXXXX.log under CTS log files

8941

UCM



• TX:

SIP/2.0 200 OK

Via: SIP/2.0/TLS 10.4.21.11:5061;branch=z9hG4bK9ed40f8dee6

a=rtpmap:97 H264/90000

a=fmtp:97 profile-level-id=42001F;packetization-mode=0;max-mbps=108000;max-fs=3600;max-br=4000

• RX:

ACK sip:[email protected]:47464;transport=tls SIP/2.0

a=rtpmap:97 H264/90000

a=fmtp:97 profile-level-id=420014;packetization-mode=0;max-mbps=36000;max-fs=1280;level-asymmetry-allowed=1

• TX: SIP REINVITE

INVITE sip:[email protected]:5061;transport=tls SIP/2.0

a=rtpmap:97 H264/90000

a=fmtp:97 profile-level-id=42000D;packetization-mode=0;max-mbps=11880;max-fs=396;max-br=4000

• RX : Post Re-Invite

SIP/2.0 200 OK

Via: SIP/2.0/TLS 10.4.22.146:47464;branch=z9hG4bK15d3420a

a=rtpmap:97 H264/90000

a=fmtp:97 profile-level-id=420014;packetization-mode=0;max-mbps=36000;max-fs=1280;level-asymmetry-allowed=1

Debugging CTS logs

Level 3.1 and VGA

Level 2.0 from 8941 and VGA

Level 1.3 from CTS

Level 2.0 from 8941 and VGA

1

2

3

4

No common Video Codec is negotiated



CTS does not support VGA and it re-INVITES to lower the resolution to w360p

8941 does not support w360p




CTS will support VGA ( in roadmap)

8941 will support w360p ( in roadmap)

Solution



Topology

• 8945 SCCP UCM SIP Trunk VCS SIP Codian MCU ,

User Experience

• Video freezes when the endpoint joins the conference

Logs to collect

• UCM SDI /SDL logs

• MCU logs

Case Study 13 : Mid Call resolution issue with 8941 while joining conference

D0: SIP Trunk

8941

VCS

MCU

UCM



Codian first sends CIF (Welcome screen), then briefly SIF (8941 video downscaled), then finally VGA (8941 video full-res)

8941 has trouble switching decode resolutions mid-stream to VGA




Workaround

Maintains VGA resolutuion

throughout the conference



Workaround

8941 connected to MCU

in VGA mode



Upgrade 8941 firmware to 9.1 (2)SR1 , Firmware adapt to dynamic resolution during mid call

Solution

Available on

cisco.com



Identifying Profile Level from H.264 SPS

while the call is active

level 2.0 in the H.264 SPS



Resolution and Profile Level mapping

Resolution Expected Profile Level

SIF : 352 x 240 1.1

CIF : 352 x 288 1.1

WCIF : 512 x 288 2.2

VGA : 640 x 480 : w360 3.0

720p : 1280 x 720 3.1

1080p : 1920 x 1080 4.0


157

Debugging Third party Payload Negotiation Issues


Case Study 14 : Polycom is not able to retrieve voicemail from Unity connection

• Topology

Polycom HDX 4000 –Third Party SIPUCM SIP Trunk Unity Connection

• Logs to Collect :

UCM SDI, SDL &Sniffer Trace


Unity connection

Polycom 4000



Debuging SIP Invite from Polycom• Comparison between Polycom and 9971 INVITE to Unity connection

PT: 101 not

negotiated

PT: 101

negotiated

Polycom Trace 9971 Phone


Root Cause


Polycom Video Endpoint does not negotiate Telephony-event in its initial INVITE


Solution

Upgrade Polycom Software version to to 3.0 available online

3.0 software advertises telephony-events


Telephony-

event

negotiated



Case Study 15 : One way Video between CTS and Polycom

CTS 1000 Polycom 4000

Topology

CTS SIP UCM Third Party SIP Polycom HDX 4000

•Logs to Collect

• UCM SDI / SDL and Sniffer Traces

UCM



Debugging one way video:200ok from polycom negotiates H.264 in 109 PT

Polycom

negotiated PT

for H.264 :109

Polycom IP



Debugging one way video : UCM ACK’s back with different payload

UCM negotiated

PT : 97

UCM IP



Different payload negotiated leads to Video codec negotiation failure




Polycom is working on addressing this issue

Solution


167

Debugging Third Party registration issues with UC Manager < 8.5



• To register any Third party Advanced SIP endpoint to UCM

Add mac address of the endpoint to UCM

Associate the endpoint with digest user and password which is created under end user

Unable to register E20 /EX90/C20 ?

Go to Advanced configuration on the endpoint and configure the following.

EnsureDefaultTransport is set to UDP or TCP towards UCM and Type is set to :Standard”

Configure loginName and Password that is set as digest user and pin under device page in UCM

Under conference 1 , ensuredegault call is set to SIP

Phone C

onfig

ura

tion


169

SIP , SCCP , H323 , BFCP Interop Call flows



H323- SIP Call Flow



H323- SCCP Call Flow



SCCP- SIP Call Flow



Typical BFCP Call Flow

Scenario : Presentation negotiated in Initial Setup

Presentation

established

3333 , 4444 , 1

Presentation

using m-

stream 12

Main Video

Channel

Presentation

Channel

Pointer to

media stream

Role of Media

stream


175

Visit the Cisco Store for

Related Titles

http://theciscostores.com

http://theciscostore.com/


Thank you.

troubleshooting unified communication

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