JMP206 : Calling Home: Enabling the IBM Sametime Softphone in ST9

JMP206 : Calling Home:
Enabling the IBM Sametime
Softphone in ST9
Keith Brooks, ThinkRite
Jeremy Sanders, ThinkRite

© 2014 IBM Corporation

Who Are These Guys?
Keith Brooks is the Social Collaboration Practice Leader for ThinkRite and a
Certified Administrator for IBM Sametime and Notes and Domino.

Keith manages a team that is responsible for providing Sametime and SUT
services to over 500,000 customers worldwide.
ThinkRite is the sole provider of SUT installations for IBM Worldwide.

Websphere (2013) and ICS (2013-2014)

Twitter/Skype: @lotusevangelist
Blog: http://blog.vanessabrooks.com
3

Who Are These Guys?
Jeremy Sanders is Chief Technical Officer of ThinkRite UK and an experienced integrator
and developer of enhancements for IBM Sametime Unified Telephony (SUT) and IBM Unified
Messaging (UM) for WebSphere Voice Response. He holds an MSc in Project Management
and achieved professional certifications in Cisco, Siemens/Unify and IBM/Lotus voice areas.
He has profound experience in integrating telephony systems and protocols.
Jeremy has worked in VoiceRite/ThinkRite for 13 years, starting as a Senior Engineer. Before
this Jeremy worked in IBM as Lead Developer for IBM UM and with another IBM business
partner installing and enhancing IBM UM.

Jeremy still occasionally works with IBM UM developers and support staff in IBM Hursley labs
but has been focused on IBM SUT for the last four years, and continues to lead SUT
installations and integrations in Europe.

4

Why Are We Here?
One word…..FREE!
Inside every downloadable copy of IBM Sametime 9 is a FREE
Repeat, FREE, product that will change your world.

We are here to show you why and how it will do this.

Please hold any questions until the end.
5

Why Are We Here?
“SUT Lite” is now called IBM Sametime Softphone (ST )
(In the ST Wiki find it under the name: Deploying SIP based calling)

It will make you and your employees lives better.
But how?

What does it do?
How can you enable this?

What else can you do with Sametime Unified Telephony? (SUT)
6

Outbound Calls to Numbers, SIP URIs or
straight from your Contacts

7

Inbound Calls with Names, Numbers,
Pictures…

8

Integrated Call History and Phonebook

9

Automated or Manual Conference
Passcodes

10

Add more Plugins to unlock more functionality

11

Introductions
Sametime 101 Class
Sametime Phone (ST )
Beyond ST 
Demo

12

Sametime 101
7.5

Know

9.0

Community Community
Server
Server (Domino)
(Domino)

Domino, Websphere
and DB2 Required

Domino, Websphere
and DB2. Linux and
Windows Required

Gateway

Gateway, Proxy

Gateway, Proxy,
Edge, TURN, Media
Manager

Gateway, Proxy,
Edge, TURN, VMCU

Entitlement, Entry,
Standard, Advanced

Communicate,
Conference and
Complete

SUT

You

8.5

Entitlement,
Entry, Standard,
Advanced

Think

8.0.2

SUT
SUT Lite

SUT,
Voice/SoftPhone

Sametime?

13

Sametime 101 – 8.5.x
8.5.1 Required the following components if you wanted JUST instant messaging and presence with meetings:
 Lotus Sametime System Console - used for managing and administering servers from a central location
 DB2®
 An LDAP directory
 Lotus Sametime Community Server (Runs on Domino)
 Lotus Sametime Meeting Server

 Lotus Sametime Proxy Server - provides an integrated web chat client and presence; required for web clients and ST browser clients
 The following components can optionally be deployed:
 Lotus Sametime Gateway - extends instant messaging to external communities
 Lotus Sametime Media Manager - provides audio and video features in the Lotus Sametime client and in meetings
 Audio-visual (AV) components provided with the Lotus Sametime Media Manager

 The Lotus Sametime Media Manager comprised three components
 SIP Proxy/Registrar – central server for audio and video clients and servers to register their location and send and receive calls
 Conference Manager - manages conference media flows
 Packet Switcher- routes audio and video data to conference participants based on detecting the active speaker

15

Sametime 101 – 9.0
9.0 Required the following components if you wanted JUST instant messaging and presence with meetings:
 IBM Sametime System Console - used for managing and administering servers from a central location
 DB2®
 An LDAP directory
 IBM Sametime Community Server (Runs on Domino)
 IBM Sametime Meeting Server

 IBM Sametime Proxy Server - provides an integrated web chat client and presence; required for web clients and ST browser clients
 The following components can optionally be deployed:
 IBM Sametime Gateway - extends instant messaging to external communities
 IBM Sametime Media Manager - provides audio and video features in the IBM Sametime client and in meetings
 Audio-visual (A/V) components provided with the IBM Sametime Media Manager

 The IBM Sametime Media Manager comprises four components
 SIP Proxy/Registrar – central server for audio and video clients and servers to register their location and send and receive calls
 Conference Manager - manages conference (and ST ) media flows
 Video MCU - enables multi-way, audio and video conferences with continuous presence and multiple client layouts
 Video Manager - manages the scaling and distribution of audio and video conferences

16

17
http://www-10.lotus.com/ldd/stwiki.nsf/xpDocViewer.xsp?lookupName=Administering+Sametime+9.0+documentation#action=openDocument&r es_title=Sametime_server_architecture_st9&content=pdcontent

18
http://www10.lotus.com/ldd/stwiki.nsf/xpDocViewer.xsp?lookupName=Administering+Sametime+9.0+documentation#action=openDocum
ent&res_title=Extending_audio_and_video_to_external_clients_and_mobile_users_st9&content=pdcontent

Sametime 101 – The Power Tools in 9.0
1. System Console Server

7. Proxy Server

2. Gateway Server

8. SIP Edge Proxy

3. Media Manager( in 4 parts) 9. TURN Server
4. Bandwidth Manager

10. Community Server / Mux

5. Meeting Server

11. LDAP Server

6. Advanced Server

12. DB2 Server
Note: ALL Components are 64bit ONLY

19

Sametime 101
1) Sametime System Console/Server:
First “server” to be installed
Use the SSC to install, configure and administer the other servers.
Larger environments this would be a stand alone server

20

Sametime 101
2) Gateway Server:
The Sametime Gateway server is used to connect Sametime clients with other instant
messaging clients. It can be a stand alone or a clustered installation.
You can, and usually do, install it securely in the network DMZ because:
– Firewall restrictions make it impossible for users from the Internet to directly access
a Sametime community server on your corporate intranet, but Internet users can
access Sametime Gateway Server in the network DMZ.
– Sametime community servers, behind the internal firewall, are accessible only over an encrypted VP
protocol.
– DB2 is behind the internal firewall, restricted by host and port access.
– LDAP is behind the internal firewall, accessible over SSL and restricted by host and port access
– Sametime Gateway Server exchanges with other instant messaging providers over SIP can be
encrypted with SSL.

Connects you to: AOL, Google Talk, and XMPP communities, Other Sametime communities
and other Sametime companies using AOL clearinghouse
21

Sametime 101
3) Media Manager:
The Sametime Media Manager is now comprised of four server components, which
can be installed on a single computer, individually on separate computers, or as a
cluster that supports fail over and high availability. They are:
1. SIP Proxy/Registrar – central service using industry standard SIP allows clients
and servers to register their locations and send and receive calls
2. Conference Manager - manages the state of audio and video calls (includes
TCSPI adapter for integration with other vendors)
3. Video MCU* - enables multi-way, audio and video conferences with continuous
presence and multiple client layouts (replaces more primitive Packet Switcher)
4. Video Manager - manages the scaling and distribution of audio and video
conferences
Works with the Sametime Bandwidth and TURN Servers (both Optional)
22

*=Linux Only

Sametime 101

3A) SIP Proxy/Registrar
Manages location services and forwards SIP messages to their destinations.
The SIP Proxy/Registrar maintains the registry between all users and their
location, and maintains the registration of conferences.
The SIP Proxy/Registrar routes all SIP messages inside Sametime. Every
voice or video message to a user goes through the SIP Proxy/Registrar.

The following components know to consult the registrar: Sametime Media
Manager, SIP-based calling, and Sametime Unified Telephony.
It requires access to LDAP.

23

Sametime 101

3B) Conference Manager
Administers all conferences, including point-to-point and multipoint.
The Conference Manager works with the client to establish a SIP session
for the call. It also hosts the internal Telephony Conferencing Service
Provider Interface (TCSPI) adapter and an optional external TCSPI
adapter. The TCSPI integrates with the Video MCUs and bridges.
The Conference Manager works with the client to establish the SIP session
for the call. The Conference Manager manages the state of audio and
video calls. All audio and video features, both one-to-one A/V chat and
multi-way A/V chats, depend on this component.

24

Sametime 101

3C) VMCU Server:
Video Multipoint Control Unit
The Video MCU provides conferencing functionality.

The main purposes of the Video MCU is to:
• Handles media and media control from Sametime 9 clients
• Routes H264 AVC (SVC base layer) and SVC video to video enabled
clients
• Routes Scalable Audio Codec (SAC) to Sametime 9 clients
• Support audio transcoding and audio mix for clients which don’t
support SAC

25

Sametime 101
3D) Video Manager
Distributes audio and video communications among the servers within
a Sametime deployment according to routing rules that you define.
The IBM Sametime Video Manager manages the scaling and distribution of audio
and video conferences, through MCU pools and cascading.
It also manages attributes for conferences, such as maximum line-rate, and the
following tasks:
– Multi-way audio and video conferencing (requires Sametime Conference)
– Multimedia transport and bandwidth control
– Call server routing based on dial plan
– Creates meeting rooms based on template
– The Video Manager cannot be clustered, but you can have multiple servers
with a load balancer in front.
26

Sametime 101
4) Bandwidth Manager:
Works with the Sametime Media Manager, controlling bandwidth used in audio and
video calls that are handled by the media manager.
You can install the bandwidth manager on an existing Sametime Media Manager,
or on a separate computer. Not managed by the Sametime System Console (SSC).
Optimizes bandwidth by calculating the call route for each call as it is initiated, and
reserving the required bandwidth for the duration of that call.

The bandwidth manager client is built into the Sametime Connect client, web client,
and embedded client, so its features are installed automatically.
The bandwidth manager is a J2EE SIP application running on IBM WebSphere®
Application Server.

You will not be able to change your mind and add the server to a cluster later!
27
You have been warned!

Sametime 101
5) Meeting Manager
Provides meeting features and online "meeting rooms" where users can present
information and share applications.
If you deploy the Sametime Media Manager, conferences can include audio and
video features as well.
The Sametime Meeting Server uses a DB2® database for storing information about
meeting room settings and schedules.

28

Sametime 101

6) Advanced Server:
Brings social networking features to the Sametime client with such
features as persistent chat rooms, broadcast messages, polling, screen
sharing, and remote machine control.
Sametime Advanced allows the ability to send and receive offline
messages, send files to a group chat, or send a folder to a contact.

The Advanced server requires a DB2 database and connection with
LDAP for authentication. Persistent chat rooms are managed and
maintained by the Advanced server itself and transcript logs in those
rooms are automatically accessible on the server.
Instant screen sharing requires you install the Meeting Server as well.
29

Sametime 101
7) Proxy Server:
Regulates communication with Sametime clients running on mobile
devices.
It’s all about PUSH.
Generally required to work with Apple (Apple’s Push Notification Server)
and Android devices (Google Cloud Messaging).

30

Sametime 101
8) SIP Edge Proxy Server:
A SIP application installed over WAS and uses the Media Manager’s SIP
Proxy/Registrar installer.
The IBM SIP Edge Proxy server connects external clients to the Sametime SIP
Proxy/Registrar server.

Both external (IBM SIP Edge Proxy IP) and internal clients (Media Manager’s SIP
Proxy/Registrar IP) receive a host name for the SIP Proxy/Registrar.
The IBM SIP Edge Proxy and the Sametime SIP Proxy/Registrar servers
communicate with each other over SIP ports.
SIP ports must be opened in the firewall in both directions and both servers
should be able to resolve the FQDN of each other.
31

Sametime 101
9) TURN Server:
Enables Sametime clients to send audio and video communications
across a NAT or firewall when direct peer-to-peer communications are not
possible.
Runs on Linux™ or Windows™ platforms only and is part of a JRE.
The TURN Server does not require WAS.
It has no dependencies on other processes or other Sametime servers.
The Sametime Connect client, Sametime Web meetings, and the Media
Manager can use the TURN server if they detect its presence.

32

Sametime 101
10) Community Server / Mux:
IBM Domino Server, Relies on Domino Clustering

Turn off all unnecessary Domino tasks
Mux Server sits in front of the Community Server for authentication and
scaling and can increase capacity significantly. No Clustering, use more.

11) LDAP Server:
IBM Domino Directory used as an LDAP repository is a popular choice
Microsoft’s Active Directory has been seen in the wild as well
Be aware different syntax and details are required for each so read the wiki
33

Sametime 101
12) DB2 Server
You can use the DB2 installation that comes with IBM Sametime, or you
can use an existing DB2 infrastructure.
The DB2 Setup wizard provides dynamic size estimates based on the
components selected during a typical, compact, or custom installation.

On Linux and UNIX operating systems, 2 GB of free space in the /tmp
directory is recommended.
Note: On Linux and UNIX operating systems, you must install
your DB2 product in an empty directory. If the directory that you have
specified as the install path contains subdirectories or files,
your DB2 installation might fail.
34

Introductions
Sametime 101 Class
Beyond ST 
Demo

35

Sametime Phone (ST)
“SUT Lite” is now called IBM Softphone,
Sametime Phone or Sametime Voice
now available in Communicate and
Complete with no additional license!

What is Sametime Phone?
A basic SIP softphone packaged with Sametime 9
A basic SIP integration from Sametime Media Manager to
an IP PBX or other SIP entity

What isn’t Sametime Phone?
A Unified Telephony solution which includes Unified
Number, multiple devices and integrations, transfers, adhoc conference calls, etc.

What is SIP?
A straight forward internet standard text based protocol
(RFC 3261) like HTTP or SMTP - but for Initiating
prolonged Sessions (Calls or Chats!)

What isn’t SIP?
Something which sends and receives audio and video (that is
RTP – Real Time Transport RFC 1889 / 3550, a binary bits and
bytes protocol using codecs like G.711)
Something which describes the audio and video to be sent (that
is SDP – Session Description RFC 3264, a text based protocol)

SIP 101 – Requests (Methods)
REGISTER

- tell a server where we are so we can receive calls

INVITE
ACK
BYE
CANCEL

- attempt to start a call ( / session / dialog / chat / …)
- 3-way handshake only used with INVITE
- ends a call ( / session / … )
- give up an attempt to start a call

OPTIONS

- check other end is there and what it can do

INFO

- mid-session/call information

MESSAGE

- instant message (session/call not required)

SUBSCRIBE

- ask for events

NOTIFY

- send event to subscriber

PUBLISH

- send event to server

UPDATE

- modifies session

REFER

- call transfer

PRACK

- provisional acknowledgement

SIP 101 - Responses
Many are just like HTTP Response Codes (e.g. 404 Not Found )
1xx Provisional (eg, 100 Trying, 180 Ringing, 183 Session Progress)
2xx Successful ( eg, 200 OK, 202 Accepted )
3xx Redirection (eg, 302 Moved Temporarily )
4xx Request Failure – eg,
–
–
–
–
–
–
–
–
–
–

401 Unauthorized / 407 Proxy Authentication Required
403 Forbidden / User Not Authorized
404 Not Found (no such user / number)
408 Request Timeout
480 Temporarily Unavailable – Channels Not Inservice
481 Call Leg/Transaction Does Not Exist
482 Loop Detected
486 Busy Here (phone may just be busy or user may have set “do-not-disturb”)
487 Request Terminated
488 Not Acceptable Here (usually a codec issue)

5xx Server Failure (eg, 501 Not Implemented, 503 Service Unavailable )
6xx Global Failure (eg, 600 Busy Everywhere, 603 Decline )

Simple Direct SIP Call
Caller (UAC)

Called (UAS)
INVITE (w or w/o SDP)
100 Trying (Provisional)
180 Ringing (Provisional)
200 OK (w SDP)
ACK (w or w/o SDP)
media (RTP)
media (RTP)
BYE
200 OK

Overview of a SIP Request
INVITE sip:jo@example.co.uk SIP/2.0
To: tester1<sip:jo@example.co.uk>
From: tester2<sip:bob@193.195.52.229>
Call-ID: fd835c@193.195.52.229
Via: SIP/2.0/UDP 193.195.52.229:5060
CSeq: 1 INVITE
Content-Type: application/sdp
Content-Length: 125

Start Line (Method / Request URI)
“command” and “to whom”

Headers – about the session

Blank Line

v=0
o=193.195.52.229 4858 0 IN IP4 193.195.52.229
s= Call from tester2
c=IN IP4 193.195.52.229
m=audio 5004 RTP/AVP 0

Body (optional) – about the media

Mandatory SIP Headers
 Method / Request URI at top (“command” and “to whom”)
 To – intended destination
 From – originator
 Call-ID – together with To (+tag added in 200) and From (+tag from
INVITE) identifies a dialog
 Cseq - the sequence number of this request
 Max-Forwards – maximum times this can be forwarded

 Via – where the message came from (and where to send the initial
responses) – can allow loop detection

Mandatory SIP Headers
INVITE sip:470000@10.1.1.12:5060 SIP/2.0

Via: SIP/2.0/TCP 10.1.6.10;branch=z9hG4bK1e2d269ab98b
From: <sip:489988@10.1.6.10>;tag=da481a1a-5547-4029-8e4c-08ae0b1dd568-30511025
To: <sip:470000@10.1.1.12>
Call-ID: 84a68180-c801ff58-7e17e-a06010a@10.1.6.10
CSeq: 101 INVITE

Max-Forwards: 70

Expected SIP Headers
 Contact – where to send subsequent responses (direct to
originator, also used in 3xx redirect response to change
destination)
 Allow – what methods are supported
 Supported – what options are supported

 Content-Length – how long the content is
 Content-Type – what the content is – eg, application/sdp,
simple-message-summary, multipart-mime

Expected SIP Headers
INVITE sip:470000@10.1.1.12:5060 SIP/2.0
To: <sip:470000@10.1.1.12>
Call-ID: 84a68180-c801ff58-7e17e-a06010a@10.1.6.10
Supported: timer,replaces
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Contact: <sip:489988@10.1.6.10:5060;transport=tcp>
Max-Forwards: 70
Content-Length: 0

Optional SIP Headers
 Remote-Party-ID – real calling number details
 P-Asserted-Identity – real calling number details (real RFC)
 Diversion – original called number info
 History-Info – original called number info (real RFC)
 Expires / Min-SE – used for session timers
 Route / Record-Route – used to stay in path
 Date / Timestamp / User-Agent (many others!)

Optional SIP Headers
INVITE sip:470000@10.1.1.12:5060 SIP/2.0
Remote-Party-ID: <sip:489988@10.1.6.10>;party=calling;screen=yes;privacy=off
To: <sip:470000@10.1.1.12>
Date: Fri, 03 Sep 2010 13:59:52 GMT
Call-ID: 84a68180-c801ff58-7e17e-a06010a@10.1.6.10
Supported: timer,replaces
Min-SE:

1800

User-Agent: Cisco-CCM5.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Contact: <sip:489988@10.1.6.10:5060;transport=tcp>
Expires: 180
Allow-Events: presence
Session-Expires:
Max-Forwards: 70
Content-Length: 0

1800

SDP Body (for a Call / Session)
Session Description Protocol:
v=0

<- v= version

o=User-Agent 2923 9482 IN IP4 10.1.1.8

<- o= media stream version details

s=-

<- s= session name

a=SIP Call

<- a= attribute

c=IN IP4 10.1.1.8

<- c= connection info including transport IP address

t=0 0

<- t= time session is active

m=audio 19144 RTP/AVP 8 0 18 101

<- m= media types and transport UDP (RTP) port

a=rtpmap: 8 PCMA/8000

<- a= attribute of media (A-law) preferred

a=rtpmap: 0 PCMU/8000

<- a= attribute of media (u-law) second choice

a=rtpmap: 18 G729/8000

<- a= attribute of media (G.729a) third choice

a=rtpmap:101 telephone-event/8000

<- a= attribute of media (DTMF)

a=fmtp:101 0-15

<- a= attribute of media (DTMF keys)

SIP INVITE with SDP Body
INVITE sip:33331@10.1.1.12:5060;lr SIP/2.0
Call-ID: 02678072b3df1b204c2aa05400
CSeq: 1 INVITE
From: sip:489988@sessim.avaya.com;tag=02678072b3df1a204c2aa05400
Record-Route: <sip:10.1.7.33:5060;lr>,<sip:10.1.7.32;lr;transport=tcp>
To: "33331" <sip:33331@sessim.avaya.com>
Via: SIP/2.0/UDP 10.1.7.33:5060;branch=z9hG4bK0303032323236363632bdd.0,SIP/2.0/TCP 10.1.7.32;psrrp
osn=2;received=10.1.7.32;branch=z9hG4bK02678072b3df1c204c2aa05400
Content-Length: 142
Content-Type: application/sdp
Contact: <sip:489988@10.1.7.32;transport=tcp>
Max-Forwards: 68
User-Agent: Avaya CM/R015x.01.0.414.0
Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH
Supported: 100rel,timer,replaces,join,histinfo
Alert-Info: <cid:external@sessim.avaya.com>;avaya-cm-alert-type=external

Min-SE: 1200
Session-Expires: 1200;refresher=uac

v=0
o=- 1 1 IN IP4 10.1.7.32
s=c=IN IP4 127.0.0.2
b=AS:64
t=0 0
m=audio 2416 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

Codec Negotiation
 One side sends SDP with all codecs supported – usually in order of
preference
 Other side chooses first codec it too supports or what it prefers
 Media only starts once this agreement has been reached
 Some IP PBXes send INVITE with no SDP and send SDP in the ACK
instead (Delayed Offer)
 183 Session Progress may contain Early Media
 To receive Early Media PRACK (Provisional ACK) support is required
if INVITE had no SDP

SIP Entities
 UAC : User Agent Client (most entities)
 UAS : User Agent Server (also most entities!)
 Proxy – for hierarchical routing (UAC and UAS)

 (Softphone) Registrar – keeps details of users’ current (IP address)
location
 (PSTN) Gateway – to Public Switched Telephone Network
 IP PBX : Back to Back User Agent (B2BUA)
 SBC : Session Border Controller : SIP Firewall / NAT workaround – to
connect 2 SIP networks
 Servers – such as Unified Messaging / Voice mail

SIP Registration with Authentication
Registrar (UAS)

User (UAC)

REGISTER
100 Trying (Provisional and Optional)
403 User Not Authorized (with encoding details)
REGISTER (with encoded password)
200 OK
…calls made/received…

REGISTER (with Expires of 0)
200 OK (or 404 User Not Found)

Sametime SIP PR Registrations
 Sametime AV – REGISTERs using Sametime user-id from Client (has no
telephone number) – used by “Call Computer”
 Sametime Phone / SUT Lite : REGISTERs using telephoneNumber from
LDAP
 Sametime Unified Telephony : REGISTERs using Unified Number with a
special softphone prefix
54

Regular SIP Proxy Inbound Call
Caller (A)

Proxy
INVITE
100 Trying

Called (B)
INVITE
100 Trying
180 Ringing

180 Ringing

200 OK

200 OK
ACK

ACK

media (RTP)
BYE
200 OK

ST Proxy Inbound Call
IP PBX (A)

INVITE (PBX SDP)
100 Trying

200 OK (ST SDP)
ACK

SIPPR

Client (B)
(internally send call
to conference focus)

MESSAGE
200 OK
Answer Call (Virtual Places)
INVITE (PBX SDP)
180 Ringing
200 OK (ST SDP)
ACK

media (RTP)
BYE
200 OK

BYE
200 OK

ST Proxy Inbound Call (Delayed Offer)
IP PBX (A)

SIPPR

INVITE
100 Trying

Client (B)
(internally send call
to conference focus)

MESSAGE
200 OK
Answer Call (Virtual Places)
INVITE
180 Ringing
200 OK (ST SDP)

200 OK (ST SDP)
ACK (PBX SDP)

ACK (PBX SDP)

media (RTP)
BYE
200 OK

BYE
200 OK

Outbound ST Call
 Requested by Sametime Client over VP (Virtual Places) protocol
 Created by Conference Manager/Focus acting as a SIP B2BUA
 Consists of Two Calls through the proxy:
–First to ST Client
–Then to the dialled number/URI
 Note that this is not the way any other softphone would make a call, which
would send an INVITE straight to the Proxy
 (Full) Sametime Unified Telephony also uses a SIP B2BUA in this way
58

ST Proxy Outbound Call
IP PBX

SIPPR

INVITE (ST SDP)
100 Trying
180 Ringing
200 OK (PBX SDP)
ACK

Client
Make Call (Virtual Places)
INVITE
180 Ringing
200 OK (ST SDP)
200 OK (ST SDP)
200 OK (ST SDP)
200 OK (ST SDP)
ACK (PBX SDP)

media (RTP)
BYE
200 OK

Ideal Troubleshooting Tool: Wireshark
 Network sniffer and network trace interpreter
 Dedicated menus and sophisticated tools included for analysing SIP/VoIP
calls
–Display calls within a trace
–Examine call flow for each call
–View SIP headers and SDP body, RTP codec used
–Even extract audio (for some codecs)
 Freely available http://www.wireshark.org/

TLS and sips:
 Transport Layer Security is the same encryption method used
in HTTPS for secure web pages
 Implementation of TLS is mandatory for SIP proxies, redirect
servers and registrars
 A sips: URI scheme (otherwise identical to the sip: scheme)
indicates that all hops between the requestor and the
resource identified by the URI must be encrypted with TLS
 Wireshark cannot (without keys and configuration) decrypt
SIP secured with TLS

SRTP
 Secure Real Time Transport Protocol for encryption audio
 Keys are exchanged in secured (TLS) SIP SDP codec
negotiations – so SRTP security depends on TLS security
 Wireshark cannot (easily) decrypt media secured with SRTP

When SIP and/or media secured…
Client-side tracing:
– rcpinstall.properties in
– C:User[name]AppDataRoamingLotusSametime.config OR
– C:notesdataworkspace.config
# use for basic audio/video session troubleshooting
com.ibm.collaboration.realtime.multimedia.phonegrid.internal.client.level=FINE

# use for ICE (STUN/TURN) troubleshooting
com.ibm.ice.level=FINE
# use for SIP troubleshooting – see sip.log in ..logs directory
com.ibm.collaboration.realtime.telephony.softphone.level=FINE

# use to enable softphone logging
com.ibm.collaboration.realtime.telephony.softphone.mfw.level=FINEST
http://pic.dhe.ibm.com/infocenter/sametime/v8r5/topic/com.ibm.help.sametime.v85.doc/trouble/trbl_client_log_trace.html

When SIP and/or media secured…
Server-side tracing on SIP PR
– Troubleshooting > Logs and trace > STMediaServer > Diagnostic Trace >
Change Log Detail Levels
*=info: com.ibm.ws.security.*=all: com.ibm.ws.sip.*=all:
com.ibm.wsspi.sip.*=all: com.ibm.ws.udp.*=all: com.ibm.sip.*=all
http://pic.dhe.ibm.com/infocenter/sametime/v8r5/index.jsp?topic=%2Fcom.ibm.help.sametime.v85.doc%2Ftrouble%2Ftrbl_av_diagtrace.html

ST Topology Overview
Sametime
Community
Server

VP

Sametime
Media Manager
Conf Mgr
B2BUA
SIP

VP
Sametime

Sametime
Media Manager
SIP PR

SIP

Client + 

SIP

SIP
Sametime
VMGR

SBC / IP PBX /
Gateway

( may be SIP)
Sametime
Bandwidth
Manager



(may
be
SIP)

PSTN

68
Design by the speaker: http://www.images-graphics-pics.com/signs/signgenerator/simple.asp?text=Sametime+Softphone+Configuration++Section&align=Center&align2=Middle&fontsize=20&
font=SignBold&color=black&transparency=255&color2=YELLOW&watermark=&move=0&move2=&parody=&rotate=&a
llow=6112&left=&right=&up=&down=&pic=Tower-Caution&x=4&y=253&w=259&h=139

Configuring IBM Sametime SoftPhone / “SUT Lite”
You have installed Sametime 9
Everything works and you are ready for more

You look for the documentation and you find some here: http://goo.gl/UHVAEy

A Zero to Hero for SUT Lite (8.5.2)? http://www.slideshare.net/jackdowning/sut-lite-client
The online course from IBM http://www-304.ibm.com/events/idr/idrevents/detail.action?meid=5128&ieid=2186
And also find this:
“The IBM Sametime Unified Telephony Lite Client is easy to deploy and does not require any additional hardware or
software over Sametime Standard. It simply requires a SIP trunk to be configured between the Sametime Media Manager
and a certified SIP environment.” – Julie Reed, Product Manager SUT

Recommended: IBM Sametime 8.5.2 SUT Lite Troubleshooting:
http://public.dhe.ibm.com/software/dw/lotus/sametime/sut/sutlitetroubleshoot.pdf

69

Deploying the ST
1.

Think about how phones will call softphones and vice-versa (the “Dial Plan”)

Users must have a telephone Number or similar field in the directory.

Can you use a prefix like 1119 in front of the user’s regular internal or external
number in the directory? Can your PBX strip off such a prefix?

Should you add a new field to the directory for the user’s softphone?

Do you just want users to just make calls but not receive them on their softphone?

What about calling video endpoints – do you want to use SIP URIs like
sip:bob@vidcon.co.com for these?

2.

Make a list or diagram of all of the SIP entities and addresses

Try to include details of how each entity can call another.

3.

Follow basic documentation for configuring SIP-based calling
http://infolib.lotus.com/resources/sametime/9.0/ST900ACD041/en/st9_access_deploy
_av_siptrunk.html#config_lite

Download and Activate License (or edit the mediaserveradmin.war/Config.jsp file)

Turn on “Allow use of SUT Lite Clients” and “Allow calls that use SIP Trunk
capability”

70

Deploying the ST
4. (Temporarily?) Turn off SIP Security, note down and configure (unsecure) Ports
Typically: Servers > Server Types > WAS Servers > STMediaServer > Ports
SIP_DEFAULTHOST (for Conference Manager) : 5063

SIP_ProxyRegHOST : 5080

SIP port for VMGR : 5060

SSC > Sametime Servers > Sametime Media Manager – Transport Protocol : TCP : 5080 (VMGR : 5060)
Audio Video Media : Disable SRTP (these settings end up in stavconfig.xml file)
Applications > Application Types > WS Enterprise Apps > IBM Lotus SIP Registrar – Security role to user/group mapping

– AllAuthenticatedUsers : Everyone
5. (Optional:) Configure Sametime with custom telephoneNumber field in LDAP
Edit authorization.xml file to change name of telephoneNumber field
6. Configuring SIP-routing rules
SSC > Sametime Servers > SIP Proxies and Registrars – Proxy Administration – New …
See the next page for examples (these end up in the proxy.xml file).

7. Restart Media Manager
See different procedures for standalone vs cluster.

71

Let’s take a Look Shall we?
Live Code Example
Please try at your office….especially when everyone is logged
in and running a meeting 

72

SIP Routing Rules – IN / OUT
eg, IP PBX will create:
INVITE sip:1119489686@[IP address]

eg, Conference Focus will create:
INVITE sip:489686@[hostname]

IP PBX sends from its own
address
(IP PBX sends to port 5080, the
SIP_ProxyRegHOST)

sip:4896.*
matches

sip:(.+)@.*

Note port 5063 (TCP)
SIP_DEFAULTHOST to
Conference Focus is used

sip:$1@10.1.7.121:5060;transpor
t=TCP

SIP Routing Rules - Priority

CM7OUT below must be evaluated before CM5OUT as it is more specific –
CM7: 4896.*
CM5: 489.*

SIP Routing Rules - Generic
See sutlitetroubleshoot.pdf

Anything not for the clientside or for trunk must be
from the IP PBX:
(?!.*;endpoint=client.*)
(?!.*;endpoint=trunk.*).*

still specify IP address
and port of Conference
Focus

Anything dialled by the
Client is destined for an
outbound SIP trunk:
.*;endpoint=trunk.*

still specify IP address
of IP PBX

SIP Routing Rules – Generic SIP URI with IP address

Works for IP addresses only (not hostnames):
Request URI
sip:.*@[0-9]{1,3}.[0-9]{1,3}.[0-9]{1,3}.[0-9]{1,3};transport=TCP

From
.*;endpoint=trunk.*

sip:(.+)@(.+);transport=TCP

sip:$1@$2
Prioritize this rule above other rules for specific numbers

SIP Routing Rules – Generic Number

Works for numbers only

Request URI
sip:[0-9]*@.*
From
.*;endpoint=trunk.*

sip:(.+)@.*

sip:$1@[your IP PBX details]
Prioritize this rule below that for generic SIP URI

ST Configuration Tips
 Trust the Documentation with a pinch of salt

– add some common sense to the documentation – for example:
• if a section doesn’t make sense (such as Configuring a custom LDAP attribute for authentication) try to work out what it
means by referring back to this presentation or cross-referencing with other documentation
• if the title of a section is “Enabling TLS encryption for SIP-based calling” but the text only shows enabling SRTP, look back at
how you disabled TLS…

 Use TCP – at least to start with

– UDP is not supported and TLS is difficult to troubleshoot and may also result in problems

 Tune Media Manager for faster restarts as you will restart it a lot!

– Set soReuseAddr and tcp_fin_timeout/TcpTimedWaitDelay as in the Tuning section of Tips and Tricks – Quick Links

 Only use one Community in your test Client

– We encountered a bug where other communities can confuse Sametime Phone (resulted in no Call window for an inbound call)

 Check new or changed rules are not lost

– always use Apply and OK for a rule and also in the main rule overview screen,
– check the proxy.xml file has the rules and check again after restarting the Media Manager

 Make sure basic network or DNS issues are not complicating matters

– test using ping from clients and/or IP PBX to the hostname of the Media Manager
– add host names to hosts files and / or set Application Servers > STMediaServer > SIP Container > Custom Properties
com.ibm.ws.sip.sent.by.host (this is in server.xml file) to IP address – so Contact header contains just IP address

78

Command Line Tracing with tshark and
trace.log

tshark -i any -R "sip or tcp.port==5080 or tcp.port==5063 or
udp.port==5080 or udp.port==5063" -d tcp.port==5080,sip -d
tcp.port==5063,sip -d udp.port==5080,sip -d udp.port==5063,sip # for
an overview (Request URI / response) … without -i any will only see connections to
other servers (not local communication with C Focus)
-w sip_packet_capture.pcap # to send to a file for later analysis
-V -o sip.display_raw_text:true –S # to view all the SIP details “live”

> filename_for_just_this_test.sip # to redirect to a file
cd /opt/IBM/WebSphere/AppServer/profiles/poc1STMSPNProfile1/logs/STMediaServer
tail –F trace.log | tee filename_for_just_this_test.log
(for tail –F on windows you can install cygwin)
79

Troubleshooting Tips
When you restart Media Manager:

- ensure you allow time for Clients to re-REGISTER – or log them out and back in yourself or
you will see
404 Not Found
- wait a while for Conference Focus to initialize and register itself too or you will see
503 Service Unavailable
When calling an IP PBX:
- 403 Forbidden usually indicates the PBX does not have a SIP trunk configured to SIPPR
or does not trust the SIPPR
- 404 Not Found usually indicates the PBX does not like the number it has received
- beware of calls looping back to the SIPPR if the Dial Plan has not been well thought out
- a sudden BYE after a call seemed to get established usually includes a Q.850 cause code
which can explain what happened – eg, Reason: Q.850;cause=3 “No route to destination”
may mean it doesn’t know the IP to send a subsequent response back to SIPPR
80

(eg, if the details in the Contact header or Record-Route header include a hostname)

Time Out for a Demo
How about live calls, video, audio and meetings? Network allowing

81

Personally made using the http://www.widgetbox.com/widget/bart-simpson-chalkboard-generator

Introductions
Sametime 101 Class
Beyond ST 
Demo

82

ST Limitations
 Cannot use multiple devices, choose from devices, use rules,
etc.

ST Limitations
 Cannot perform PBX-like (or even other vendor softphonelike) actions like transfers

 Cannot add users to existing
calls (ad-hoc conferencing)

Other Limitations
 Must restart Media Manager whenever change SIP routing configuration
 SIP routing configuration by Regex is not very intuitive to Telephony
Administrators (is more intuitive to programmers)
 Can only transform/use numbers in Request-URI - cannot use numbers in
any other header such as To, Diversion, History-Info
 Extremely limited set of configuration options for interoperability tweaking
 Does not support G.729 codec (frequently used for VoIP over WAN)

85

Alternatives and Additions
 Full Sametime Unified Telephony –sophisticated B2BUA
“Telephony Control Server” (TCS) controlled by Computer
Supported Telecommunications Applications (CSTA) protocol
by the “Telephony Application Server” (TAS)
 Other vendor integrations with Sametime – client-side plugins
and/or server-side using Telephony Conferencing Server
Provider Interface (TCSPI)
 Voice mail / Unified Messaging plugins (these may include
functionality similar to the above for one-number/transfers)

SUT Topology Overview
Sametime
Community
Server

VP

T
A
S

T
C
CSTA
S
SIP

VP
Sametime

Sametime
Media Manager
SIP PR

SIP

Client + 

SBC / IP PBX /
Gateway

SIP
Sametime
VMGR

( may be SIP)
Sametime
Bandwidth
Manager



(may
be
SIP)

PSTN

What Can Full SUT Do?
IBM Sametime Unified Telephony adds more features for making, receiving, and managing
telephone calls.
In addition to the calls and video features of IBM Sametime Communicate, with SUT you can:
 Look at your Sametime contact list to see who is available for calls.
 Make and receive calls through any of your preferred devices, including:
– Your Sametime client.
– Traditional telephones, such as your office or home phone.
– Mobile devices.
 Route incoming calls to other devices or telephone numbers, depending on your location or other
criteria. Your SUT phone number always displays, so your personal numbers are never exposed.
 Switch a call to a different device, without interrupting the call.
 Call people inside or outside of your organization, even people who do not have SUT
 Get notified whenever someone is calling you.

 Begin ad hoc conferences with contacts not in your organization.
http://www-10.lotus.com/ldd/stwiki.nsf/dx/Get_started_with_calls_stu9

Sametime Unified Telephony







Sametime Softphone
When configured via SIP trunk to backend
telephony or video infrastructure
















“Off hook” presence status
Place / receive calls from the Sametime 9
Connect client.
Call video endpoints or video MCUs
Call telephone numbers or conference bridges
Within a call: mute/unmute, raise/lower volume,
start/stop video, leave call, hold/resume
Other features: Click to call, dial via Quickfind or
Dial Pad, view call history










Single number reaches you on any device
Intelligent Incoming call rules & routing
Multiple device support
Move an in progress call between devices
Visual audio conferencing, drag & drop
Moderator conference controls
Transfer, merge calls
Can support multiple PBXs to create a
seamless UC environment
Support includes legacy TDM PBXs

“Off hook” presence status
Place / receive calls from the Sametime 9
Connect client.
Call video endpoints or video MCUs
Call telephone numbers or conference bridges
Within a call: mute/unmute, raise/lower
volume, start/stop video, leave call,
hold/resume
Other features: Click to call, dial via Quickfind
or Dial Pad, view call history

ST vs SUT features
Feature

ST

SUT

Point-to-point calls with softphone

Y

Y

Multipoint calls (3 or more participants, ad-hoc conferencing)

N

Y

Visual audio conferencing with moderator controls

N

Y

PBX features: Call transfer, hold, merge

N

Y

List of devices which can be selected

N

Y

User rules for devices (location-/ presence-/ time-/ caller- based)

N

Y

Single number service for incoming calls using the above

N

Y

Seamlessly move calls from one device to another

N

Y

Integration with multiple PBXes, Video/Conference solutions

N

Y

Support for G.729 codec

N

Y

Supports dialling SIP URLs as well as numbers

Y

N

Telephony presence icon

Y

Y

(Telephony presence for ST has been added for Sametime 9, it was not available in original SUT-Lite)

Comparisons – Part1

http://www-01.ibm.com/software/lotus/products/sametime/telephony.html

Comparisons – Part 2

http://www-01.ibm.com/software/lotus/products/sametime/telephony.html

SUT Interoperability

http://www-10.lotus.com/ldd/stwiki.nsf/dx/SUT_Interoperability_Testing_Program

SUT-Lite Interoperability

http://www-10.lotus.com/ldd/stwiki.nsf/dx/Sametime_Unified_Telephony_Lite_Interoperability_Testing_Program

SUT vs ST  Dial Plan

 ST  dial plan : SIPPR rules
– regular expressions, normally two per “SIP trunk”
– order of the rules is important
 SUT dial plans
– easier to understand numbers and lengths
– order not important (always shown in numeric order)
– modular & powerful like a PBX: SIP endpoints (trunks),
routes to these, destinations which can have more than
one route, destination codes and prefix access codes
 “Configurator” configures SUT dial plans with minimum effort

ST  Tips and Tricks – Quick Links
 sutLiteNumberMatcher – regular expression to send call to SUT-Lite
instead of TCSPI (for other vendor video conferencing integration)
– http://social-collaboration.blogspot.co.uk/2012/02/unnoticed-parameter-ofsametime-852.html

 videoCallNumberMatcher – regular expression to start a video call
automatically when dialed
– http://social-collaboration.blogspot.co.uk/2012/05/unnoticed-parameter-ofsametime-852.html
 Sametime 9 Mobile and SUT Lite
– http://social-collaboration.blogspot.co.uk/2013/12/ibm-sametime-9-mobile-andsut-lite.html

 Regular expressions
– http://social-collaboration.blogspot.co.uk/2013/12/regex-sip-uri.html
– http://www.zytrax.com/tech/web/regex.htm
 SIPPR WAS Tuning
– http://pic.dhe.ibm.com/infocenter/wasinfo/v6r0/index.jsp?topic=%2Fcom.ibm.w
ebsphere.express.doc%2Finfo%2Fexp%2Fae%2Ftprf_tunelinux.html
– https://pic.dhe.ibm.com/infocenter/wasinfo/v6r1/index.jsp?topic=%2Fcom.ibm.
websphere.base.doc%2Finfo%2Faes%2Fae%2Frrun_chain_tcpcustom.html

 Troubleshooting
– http://www10.lotus.com/ldd/stwiki.nsf/dx/Troubleshooting_Sametime_Unified_Telephony_
Lite
– http://wiki.wireshark.org/VoIP_calls
– http://cnes.com/causecodes.html

 Video/MCU Architecture and Integration
– http://social-collaboration.blogspot.co.uk/2013/12/sip-trunks-video-mcu-dialinout-with.html
– http://social-collaboration.blogspot.co.uk/2013/12/ibm-sametime-9-video-mcusvc-bandwidth.html

Other References
1.

Sametime Wiki: http://www-10.lotus.com/ldd/stwiki.nsf

2.

Detailed system requirements for Sametime and Sametime Unified Telephony, Ver. 7-9:
http://www-01.ibm.com/support/docview.wss?uid=swg27007792

3.

List of all files needed to Download: http://www-01.ibm.com/support/docview.wss?uid=swg24035249

4.

IBM Sametime Unified Telephony Lite (SUT Lite) Self Paced Online Course from IBM:
http://goo.gl/OMk0mT

5.

SUT Lite PDF Library (We have the PDF’s if the page disappears): http://goo.gl/YrKQgY

6.

From Zero to Hero – Sametime 8.5.2 SUT Lite: http://goo.gl/j0gukP

7.

IBM Sametime 8.5.2 SUT Lite Troubleshooting:
http://public.dhe.ibm.com/software/dw/lotus/sametime/sut/sutlitetroubleshoot.pdf

8.

IBM Sametime 8.5.2 SUT Lite Configuration: http://public.dhe.ibm.com/software/dw/lotus/sametime/st852/sutlite852_config.pdf

9.

IBM Sametime 8.5.2 Administration Guide on Amazon.com: http://goo.gl/tpYkQb

99

Sametime at Connect 2014 – 35 Sessions!
SUNDAY: 1:30 - 3:30 JMP204 IBM ST9 Deployment Workshop, Dolphin S Hemi 1
4:00 - 6:00 JMP205 Step by Step IBM ST9 Web Integration and Customization,Dolphin S Hemi 2
MONDAY - 5:00-6:00 KEY105 IBM Sametime Roadmap, Dolphin N Hemi A-C
TUESDAY - 1:30 – 2:30 BP501 Building and Deploying Custom IBM ST Connect Client
Installations, Dolphin N Hemi D
WEDNESDAY - 11:15-12:15 ID304 IBM ST9 Voice and Video Deployment, Dolphin N Hemi D

4:30-6:15 SHOW401 Taking IBM Sametime Mobile, Swan Osprey 1-2
THURSDAY – 8:15 – 9:15 ID306 keep Calm and Call On! IBM ST Communicate Softphone,
Swan Pelican 1-2
10:00-11:00 ID302 Upgrading and Migrating to IBM ST9, Dolphin N Hemi E
10:00-11:00 ID301 IBM ST9 Voice and Video: Roadmap for Tomorrow,
Swan Mockingbird 1-2
100

 Access Connect Online to complete your session surveys using any:
– Web or mobile browser
– Connect Online kiosk onsite

Get out there and enjoy Connect 2014!
Be Social and talk to people!
101

Acknowledgements and Disclaimers
Availability. References in this presentation to IBM products, programs, or services do not imply that they will be available in all countries in which IBM operates.
The workshops, sessions and materials have been prepared by IBM or the session speakers and reflect their own views. They are provided for informational purposes only, and are neither
intended to, nor shall have the effect of being, legal or other guidance or advice to any participant. While efforts were made to verify the completeness and accuracy of the information
contained in this presentation, it is provided AS-IS without warranty of any kind, express or implied. IBM shall not be responsible for any damages arising out of the use of, or otherwise
related to, this presentation or any other materials. Nothing contained in this presentation is intended to, nor shall have the effect of, creating any warranties or representations from IBM or
its suppliers or licensors, or altering the terms and conditions of the applicable license agreement governing the use of IBM software.
All customer examples described are presented as illustrations of how those customers have used IBM products and the results they may have achieved. Actual environmental costs and
performance characteristics may vary by customer. Nothing contained in these materials is intended to, nor shall have the effect of, stating or implying that any activities undertaken by you
will result in any specific sales, revenue growth or other results.

© Copyright IBM Corporation 2014. All rights reserved.
 U.S. Government Users Restricted Rights - Use, duplication or disclosure restricted by GSA ADP Schedule Contract with IBM Corp.
 IBM, the IBM logo, ibm.com and Lotus, Websphere, DB2 are trademarks or registered trademarks of International Business Machines Corporation in the United States, other countries,
or both. If these and other IBM trademarked terms are marked on their first occurrence in this information with a trademark symbol (® or ™), these symbols indicate U.S. registered or
common law trademarks owned by IBM at the time this information was published. Such trademarks may also be registered or common law trademarks in other countries. A current list of
IBM trademarks is available on the Web at “Copyright and trademark information” at www.ibm.com/legal/copytrade.shtml
Other company, product, or service names may be trademarks or service marks of others.

102

JMP206 : Calling Home: Enabling the IBM Sametime Softphone in ST9

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