© 2014 IBM Corporation

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

Keith Brooks, ThinkRite

Jeremy Sanders, ThinkRite

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Introductions

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

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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.

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Who Are These Guys?

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)

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7

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

8

Inbound Calls with Names, Numbers, Pictures…

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Integrated Call History and Phonebook

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Automated or Manual Conference Passcodes

Add more Plugins to unlock more functionality

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12

Introductions

Sametime 101 Class

Sametime Phone (ST )

Beyond ST

Demo

Sametime 101

Think

You

Know

Sametime?

13

7.5 8.0.2 8.5 9.0

Community Server (Domino)

Community Server (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

Entitlement, Entry, Standard, Advanced

Communicate, Conference and Complete

SUT SUTSUT Lite

SUT, Voice/SoftPhone

Sametime 101 - Today

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Sametime 101 – 8.5.x8.5.1 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

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Sametime 101 – 9.09.0 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

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Sametime 101 – The Power Tools in 9.0

1. System Console Server

2. Gateway Server

3. Media Manager( in 4 parts)

4. Bandwidth Manager

5. Meeting Server

6. Advanced Server

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7. Proxy Server

8. SIP Edge Proxy

9. TURN Server

10. Community Server / Mux

11. LDAP Server

12.DB2 Server

Note: ALL Components are 64bit ONLY

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

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Sametime 1012) 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

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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.

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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.

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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 

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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.

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Sametime 1014) 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! You have been warned!

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Sametime 1015) 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.

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Sametime 1016) 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.

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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).

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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.

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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.

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Sametime 10110) 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

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Sametime 10112) 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.

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35

Introductions

Sametime 101 Class

Sametime Phone (ST )

Beyond ST

Demo

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!

FREE

!

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, ad-hoc 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 - attempt to start a call ( / session / dialog / chat / …)ACK - 3-way handshake only used with INVITEBYE - ends a call ( / session / … )CANCEL - 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 eventsNOTIFY - send event to subscriberPUBLISH - send event to serverUPDATE - modifies sessionREFER - call transferPRACK - provisional acknowledgement

SIP 101 - ResponsesMany 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

Called (UAS)

INVITE (w or w/o SDP)

100 Trying (Provisional)

180 Ringing (Provisional)

200 OK (w SDP)

ACK (w or w/o SDP)

BYE

200 OK

Caller (UAC)

media (RTP) media (RTP)

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

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

Start Line (Method / Request URI)

“command” and “to whom”

Headers – about the session

Blank Line

Body (optional) – about the media

Overview of a SIP Request

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

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

INVITE sip:470000@10.1.1.12:5060 SIP/2.0

Expected SIP HeadersContact – 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

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

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

INVITE sip:470000@10.1.1.12:5060 SIP/2.0

Optional SIP HeadersRemote-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 HeadersVia: SIP/2.0/TCP 10.1.6.10;branch=z9hG4bK1e2d269ab98b

Remote-Party-ID: <sip:489988@10.1.6.10>;party=calling;screen=yes;privacy=off

From: <sip:489988@10.1.6.10>;tag=da481a1a-5547-4029-8e4c-08ae0b1dd568-30511025

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: 1800

Max-Forwards: 70

Content-Length: 0

INVITE sip:470000@10.1.1.12:5060 SIP/2.0

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 BodyCall-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

INVITE sip:33331@10.1.1.12:5060;lr SIP/2.0

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

REGISTER

100 Trying (Provisional and Optional)

403 User Not Authorized (with encoding details)

User (UAC) Registrar (UAS)

REGISTER (with encoded password)

100 Trying (Provisional and Optional)

200 OK

REGISTER (with Expires of 0)

100 Trying (Provisional and Optional)

200 OK (or 404 User Not Found)

…calls made/received…

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

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Regular SIP Proxy Inbound Call

INVITEINVITE

100 Trying100 Trying

180 Ringing180 Ringing

200 OK200 OK

ACKACK

BYE

200 OK

ProxyCaller (A) Called (B)

media (RTP)

ST Proxy Inbound Call

INVITE (PBX SDP)

INVITE (PBX SDP)

100 Trying

180 Ringing200 OK (ST SDP)

200 OK (ST SDP)ACK

ACK

BYE200 OK

SIPPRIP PBX (A) Client (B)

Answer Call (Virtual Places)

BYE200 OK

media (RTP)

SIPPR rules

applied here MESSAGE200 OK

(internally send call to

conference focus)

ST Proxy Inbound Call (Delayed Offer)

INVITE

INVITE

100 Trying

180 Ringing200 OK (ST SDP)

200 OK (ST SDP)ACK (PBX SDP)

BYE200 OK

SIPPRIP PBX (A) Client (B)

MESSAGE200 OK

Answer Call (Virtual Places)

BYE200 OK

media (RTP)

ACK (PBX SDP)

SIPPR rules

applied here

(internally send call to

conference focus)

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

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ST Proxy Outbound Call

180 Ringing 200 OK (ST SDP)

200 OK (PBX SDP)ACK ACK (PBX SDP)

BYE200 OK

SIPPRIP PBX Client

INVITE180 Ringing200 OK (ST SDP)

Make Call (Virtual Places)

INVITE (ST SDP)

media (RTP)

100 Trying

200 OK (ST SDP)

200 OK (ST SDP)

SIPP

R ru

les

appl

ied

here

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/

Wireshark (Inbound Call)

Wireshark (Outbound Call)

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]\AppData\Roaming\Lotus\Sametime\.config OR– C:\notes\data\workspace\.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

Media Manager

SIP PR

SBC / IP PBX /

GatewayPSTN

Sametime

Client +

Sametime

Bandwidth

Manager

Sametime

Community

Server

SIP SIP(may

be SIP)

( may be SIP)

VP

VP

Sametime

Media Manager

Conf Mgr

B2BUA

SIP

Sametime

VMGR

SIP

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39

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 ST1. 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 callinghttp://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 ST4. (Temporarily?) Turn off SIP Security, note down and configure (unsecure) PortsTypically: 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 LDAPEdit authorization.xml file to change name of telephoneNumber field

6. Configuring SIP-routing rulesSSC > 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

sip:

$1@10.1.7.121:5060;transport=TCP

Note port 5063 (TCP)

SIP_DEFAULTHOST to Conference

Focus is used

sip:(.+)@.*

sip:4896.*

matches

eg, Conference Focus will create:

INVITE sip:489686@[hostname]

eg, IP PBX will create:

INVITE sip:1119489686@[IP address]

IP PBX sends from its own

address

(IP PBX sends to port 5080, the

SIP_ProxyRegHOST)

SIP Routing Rules - Priority

CM7OUT below must be evaluated before CM5OUT as it is more specific –

CM7: 4896.*

CM5: 489.*

SIP Routing Rules - GenericSee sutlitetroubleshoot.pdf

Anything not for the client-side

or for trunk must be from the IP

PBX:

(?!.*;endpoint=client.*)

(?!.*;endpoint=trunk.*).*

Anything dialled by the Client is

destined for an outbound SIP

trunk: .*;endpoint=trunk.*

still specify IP address and

port of Conference Focus

still specify IP address of

IP PBX

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 SIP URI with IP address

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

SIP Routing Rules – Generic Number

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 address78

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 TipsWhen 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

(eg, if the details in the Contact header or Record-Route header include a hostname)80

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

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

82

Introductions

Sametime 101 Class

Sametime Phone (ST )

Beyond ST

Demo

ST Limitations

Cannot use multiple devices, choose from devices, use rules, etc.

ST Limitations

Cannot perform PBX-like (or even other vendor softphone-like) 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

Media Manager

SIP PR

SBC / IP PBX /

GatewayPSTN

Sametime

Client +

Sametime

Bandwidth

Manager

Sametime

Community

Server

SIP(may

be SIP)

( may be SIP)

VP

VP

T

C

S

SIP

TA

S CSTA

Sametime

VMGR

SIP

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 Softphone

Sametime Unified Telephony

“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

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

ST vs SUT featuresFeature 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 PlanST 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-of-sametime-852.html

videoCallNumberMatcher – regular expression to start a video call automatically when dialed

– http://social-collaboration.blogspot.co.uk/2012/05/unnoticed-parameter-of-sametime-852.html

Sametime 9 Mobile and SUT Lite– http://social-collaboration.blogspot.co.uk/2013/12/ibm-sametime-9-mobile-and-

sut-lite.html

ST Tips and Tricks – Quick Links

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.websphere.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

ST Tips and Tricks – Quick Links

Troubleshooting– http://www-10.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-dial-ino

ut-with.html– http://social-collaboration.blogspot.co.uk/2013/12/ibm-sametime-9-video-mcu-s

vc-bandwidth.html

Other References1. 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

© 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

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.