Post on 14-Apr-2015
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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
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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.
BRKEVT-3304
Session Logistics
Come see me after session to chat and/or leave your
name, email, and question(s) so I can get back to you.
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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
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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
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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
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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
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Deployment Overview
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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
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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
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Log collection from VCS , VCS endpoints & UC Manager
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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
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• 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
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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
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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
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UC Manager Log Collection
Wireshark Capture
via UCM CLI
2
3
Cli to Capture
pcap trace
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UC Manager Integration with Cisco Telepresence VCS
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Case Study 1 : 9971 gets fast busy and the call does not get connected
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• 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
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• SIP Trunk Failure
What’s the Root Cause ?
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Debugging SIP Trunk Failure
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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
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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
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Debugging SIP Trunk Failure
SIP OPTIONS
PING from VCS
OPTIONS PING
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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
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Debugging One way Video
INVITE from UCM to VCS
200ok from VCS to UCM: Payload 97 for H264 MIME type
PT:97
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Debugging One way Video
PT:105
UCM IP
UCM ACK’s back to VCS with 105 Payload for H264 MIME Type
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• Dynamic Payload Negotiation Issue
What’s the Root Cause ?
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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
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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
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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
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Cisco Telepresence VCS Integration with Unity Connection voicemail
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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
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SIP Exchange between UCM and VCS INVITE from VCS to UCM
Request-Line: INVITE sip:28100@10.4.21.11SIP/2.0
Invite from VCS
without Diversion
header
Call Forward Scenario
in UCM case
Diversion Header
Uconn Pilot Number
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SIP Exchange between UCM and Unity connection
• Invite from UCM towards unity connection
Request-Line: INVITE sip:28100@10.4.21.49:5060 SIP/2.0
VCS Case :
Invite from UCM to
Unity Connection
UCM Case :
Invite from UCM to
Unity connection
Subscriber
DN
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Cisco Telepresence VCS does not support diversion header
What’s the Root Cause ?
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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
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Request-Line: INVITE sip:59000@10.4.21.49 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
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INVITE from VCS to UCM
Request-Line: INVITE sip:28100@10.4.21.11 SIP/2.0
What really happened in UCM failure case ?
Request URI and TO Header across SIP Trunk
VCS IP
Fwd DN
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Request sent out from UCM towards Unity connection
Request-Line: INVITE sip:28100@10.4.21.49:5060 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
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• 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
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UC Manager Session Layer Bandwidth
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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>
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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
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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
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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
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UCM 8.5 : Region Bandwidth set to 384 Kbps
200ok from UCM to E20
No AS
parameter in
the answer
No TIAS for
Audio m-line
384000 – 64000 = 320000
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UCM 8.5 : Region Bandwidth set to 384 Kbps
E20 Screenshot
E20 channel rate falls
to very low bit rate
degrading video qualityWill experience choppy or black video on E20
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Comparing with UCM 8.6 traces : Region Bandwidth set to 384 kbps
Initial INVITE from E20 to UCM
Default call
set to 1152
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Comparing with UCM 8.6 traces : Region Bandwidth set to 384 kbps
200ok from UCM to E20
UCM Indicates
Region
Bandwidth
TIAS: 64000
for G711
384000 – 64000 = 320000 for Video
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Comparing with UCM 8.6 traces : Region Bandwidth set to 384 kbps
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
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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
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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
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UC Manager BFCP Interoperability
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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
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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
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Debugging Sniffer trace
Ex90 Negotiates
UDP/BFCP
Initial Invite from EX90 to UCM
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Debugging Sniffer trace
Invite from UCM to VCS
No UDP/BFCP app
line in SDP
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What’s the Root Cause ?
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
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What’s the Root Cause ?
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
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VCS Default Communication Manger Profile sets SIP UDP/BFCP filter to “ON”
Is there any Config on VCS Side to enable BFCP ?
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when it is “ON” , VCS does TCP/BFCP Which is not supported with UC Manager and not the industry preferred choice
Is there any Config on VCS Side to enable BFCP ?
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Is there any Config on VCS Side to enable BFCP ?
Create a CUSTOM Profile and set the SIP UDP/BFCP filter mode to “OFF”
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Working Config
SIP Profile under
SIP Trunk is enabled
with BFCP
After enabling “Allow Presentation Sharing using BFCP” in SIP profile
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Working Config
UCM Negotiates
UDP/BFCP App
line
After enabling “Allow Presentation Sharing using BFCP” in SIP profile
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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
BFCP App: Non-Secure
Audio: Secure
Main Video: Secure
Presentation Video: Secure
BFCP App: Non-Secure
Service Parameter
True
True
False
Security Status of Call
Secure
Non-Secure
Non-Secure
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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
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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
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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:26002@10.4.21.11>
Remote-Party-ID: "Polycom-VSX7000-H323-26002" <sip:26002@10.4.21.11>;party=calling;screen=yes;privacy=off
180 Ringing had (no display name) –
P-Asserted-Identity: <sip:26002@10.4.21.11>
Remote-Party-ID: <sip:26002@10.4.21.11>;party=called;screen=yes;privacy=off
Remote party
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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:51991@abi.com,
Call-ID=f75faf53903b3170@10.4.22.107"
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: f75faf53903b3170@10.4.22.107
CSeq: 101 UPDATE
From:
<sip:26002@abi.com>;tag=109550~78e066fc-a56b-45ac-83ce-0a62eb5c06b2-8124
2816
To: <sip:51991@abi.com>;tag=bae9771a1d8e8596
Server: TANDBERG/4099 (X5.2)
Debugging 481 message
Response from VCS to
UCM
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• 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:26002@abi.com,
Call-ID=f75faf53903b3170@10.4.22.107"
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: f75faf53903b3170@10.4.22.107
CSeq: 100 INVITE
From: <sip:51991@abi.com>;tag=bae9771a1d8e8596
To:
<sip:26002@abi.com>;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
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VCS does not support the UPDATE before answer, or PRACK (Provisional response Acknowledgment)
What’s the Root Cause ?
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Configure Custom Profile in VCS version < 6.0 version to strip SIP UPDATE message
Solutions
SIP UPDATE
being stripped off
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Solutions
VCS 6.1 Version ignores or suppresses SIP UPDATE messages
VCS will start supporting SIP UPDATE header (in roadmap)
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Cisco Telepresence Native Interop
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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.
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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
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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
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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
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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
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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
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UC Manager Integration with Cisco Telepresence MCU
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MCU Conference Overview
E20-SIPC20-SIP
Ex90-SIP
CUCILync
CIUS-SIP
MXP 1700 SIP
MCU
Intercluster SIP Routing
UCMUCM
CUPC
E20
C20
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• MCU is added to UCM via Media Resources Conference Bridge as Cisco Telepresence MCU under
How do we add MCU to UCM ?
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Registration Status
MCU
Registered
Http
credentials of
MCU
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• MCU resource is added to MRG and then MRGL
How do we add MCU to UCM ?
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• MCU contained MRGL is configured as part of phone MRGL
How do we add MCU to UCM ?
Phone MRGL
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MCU Side Config
Media Port Reservation was changed to "Enabled“
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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"
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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
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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
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Debugging Port Usage on Codian
Non Video
phone
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Debugging MCU Logs Initial DO Invite from
UCM to MCU
MCU offers both
Audio and Video
Caps for Initial late
media
Video port Allocated
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UCM will always include video in the ACK to the Cisco Telepresence MCU because the UCM treats the MCU as pure video bridge
What’s the Root Cause ?
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• 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
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UC Manager will request Video or Audio specific port based on the phone capability ( is in the roadmap)
Alternate Solution
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UC Manager and VCS Multiway Conference
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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
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VCS and MCU Side config
Conference alias
MCU Conference
alias config
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UCM Side Routing config
REFER from VCS
to UCM when
multiway is
invokedRHS matches
TLD and SIP
Route pattern
UCM & VCS Domain : abi.com
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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)
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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
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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
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One of the Root Cause
Redirect by Application under SIP Profile of the SIP trunk is Unchecked
Ensure Redirect by
Application is checked
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With Redirect by Application checked
Solution via redirect by application
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With Redirect by Application checked
VCS sends 302 with multiway factory number to join the bridge
Solution via redirect by application
VCS to UCM : 302
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With Redirect by Application checked
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
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Active Multiway Conference
Active Registration
All 3 participants are in Multiway conference
MCU Screenshot
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108
UC Manager Integration with ISR Conference Bridge
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• 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
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UCM Config : Heterogeneous & Homogenous Mode
Adding New
conference
Bridge
Added to UCM via Media Resource Conference Bridge
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Registration Status with UCM
Conference
Bridge
registered
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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
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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
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Cisco Telepresence endpoints Native registration with UC Manager
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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
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Debugging EX/C-Series 90 Registered Native to UCM
SIP
TFTP
UCMURI
From EX/C-Series Web Interface : http://<ex/c-series_ip>
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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.
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Troubleshooting Summary
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…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
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• 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
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APPENDIX
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Endpoint feature and Interop Summary
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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
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UCM and Cisco Telepresence VCS Video / Audio Security
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• 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
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Configuring Security on VCS Side
SIP Mode Settings
Active on 5061 Port
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Configuring Security on VCS Side
Upload Private and Server Certificates
Register Secure
endpoint with VCS
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UCM Side Config
1
2
3Should match
VCS Cert x.509
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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
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Determining Security from UCM TracesUCM Invite
VCS
Response
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Additional Case Studies
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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
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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
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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
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• 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:21989@10.4.21.11>;tag=a3c7d054a3fba9d0^M
To: <sip:8333780014@10.4.21.11>;tag=697~78e066fc-a56b-45ac-83ce-0a62eb5c06b2-75907411^M
Date: Tue, 10 Aug 2010 18:37:10 GMT^M
Call-ID: 768c980444587620@10.4.22.182^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
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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
What’s the Root Cause ?
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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
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• 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
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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
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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
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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
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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
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: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
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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
m=audio 31490 RTP/AVP 0 8 18
102 9 116 124 101
b=TIAS:64000
m=video 31604 RTP/AVP 126 97
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.
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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
m=audio 16616 RTP/AVP 9 103
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
m=audio 31490 RTP/AVP 0 8 18
102 9 116 124 101
b=TIAS:64000
m=video 31604 RTP/AVP 126 97
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.
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Mid call Negotiation Issues
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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
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• 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:20065@10.4.22.146:47464;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:20048@10.4.21.11:5061;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
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CTS does not support VGA and it re-INVITES to lower the resolution to w360p
8941 does not support w360p
What’s the Root Cause ?
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CTS will support VGA ( in roadmap)
8941 will support w360p ( in roadmap)
Solution
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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
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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
What’s the Root Cause ?
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Workaround
Maintains VGA resolutuion
throughout the conference
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Workaround
8941 connected to MCU
in VGA mode
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Upgrade 8941 firmware to 9.1 (2)SR1 , Firmware adapt to dynamic resolution during mid call
Solution
Available on
cisco.com
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Identifying Profile Level from H.264 SPS
while the call is active
level 2.0 in the H.264 SPS
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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
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Debugging Third party Payload Negotiation Issues
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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
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Unity connection
Polycom 4000
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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
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Root Cause
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Polycom Video Endpoint does not negotiate Telephony-event in its initial INVITE
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Solution
Upgrade Polycom Software version to to 3.0 available online
3.0 software advertises telephony-events
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Telephony-
event
negotiated
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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
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Debugging one way video:200ok from polycom negotiates H.264 in 109 PT
Polycom
negotiated PT
for H.264 :109
Polycom IP
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Debugging one way video : UCM ACK’s back with different payload
UCM negotiated
PT : 97
UCM IP
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Different payload negotiated leads to Video codec negotiation failure
What’s the Root Cause ?
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Polycom is working on addressing this issue
Solution
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Debugging Third Party registration issues with UC Manager < 8.5
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• 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
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SIP , SCCP , H323 , BFCP Interop Call flows
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H323- SIP Call Flow
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H323- SCCP Call Flow
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SCCP- SIP Call Flow
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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
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