IBM Sametime 9 Mobile and SUT Lite

[UPDATE] No more required for latest releae of Sametime Mobile.

This morning, I have done tests on Sametime Mobile. Sametime 9 Mobile includes SUT Lite capabilities. It is enabled via the av.allowSIPTrunking policy being true and assumes SUT-Lite trunk is set up on the server side. Making phone calls using Sametime Voice on mobile, by entering a phone number on keypad will be an SUT-Lite call if SUT-Lite is enabled.  

BUT, Sametime 9 Mobile has a specific behavior. A "+" sign is automatically added in front of the number you dial. So, my classic rules and my substitution patterns don't work for Sametime mobile

So, I have added a specific rule:

RequestURI is "sip:\+12.*" to be able to recognize sip:+12xxxx
And RequestURI pattern sip:\+(.+)@.* to remove the "+" sign before sending it to the external SIP Server.

From the Sametime trace, you can see something like thats:
[12/20/13 3:09:01:501 MST] 000000c1 proxy         3 ProxyHandler handleMAddr ENTRY  {sip:+1250@sametime9.ucalab.polycom.com}
[[12/20/13 3:09:01:501 MST] 000000c1 routing       3 RoutingCondition matches ENTRY  {Condition type: Method, value: .*} {INVITE}
[12/20/13 3:09:01:501 MST] 000000c1 routing       3 RoutingCondition matches RETURN  {true}
[12/20/13 3:09:01:501 MST] 000000c1 routing       3 RoutingCondition matches ENTRY  {Condition type: RequestURI, value: sip:\+12.*} {sip:+1250@sametime9.ucalab.polycom.com}
[12/20/13 3:09:01:501 MST] 000000c1 routing       3 RoutingCondition matches RETURN  {true}

Now, my Sametime 9 Mobile client can call audio endpoints though SIP trunk.

SIP trunks, Video MCU, Dial in/out with IBM Sametime 9

SIP trunk are very important in Telephony and Video world. With IBM Sametime 8, in order to enable SIP trunk capabilities, it was required to buy IBM® Sametime® Unified Telephony Lite. This
licence enables you to place calls to and receive calls from internal or external phone numbers, video conferencing systems or video conferencing endpoints on your desktop.

With the new licensing model of IBM Sametime 9, this license is now free of charge... It is included in almost all bundles (Communicate and Complete) - link

I forecast that almost all IBM Sametime deployments will involve SIP trunks, because it opens IBM Sametime to a entirely new world:

• Device-based dialing for Sametime softphone functionality
• Call non-Sametime users — reach people on any device including Polycom RealPresence audio/video endpoints 
• Provides rich remote collaboration experience with video—reducing productivity loss, expense of travel, and telephony costs

 

SUT Lite call to a video endpoints from IBM Sametime 9 Client

In addition to enable licence in the IBM Sametime System Console, you need to define rules in Sametime SIP Proxy.
You can find two excellent guide to do it:

• From Zero to Hero - Sametime Unified Telephony Lite: link 
• IBM Sametime 8.5.2 Unified Telephony Lite Client troubleshooting guide: link 

These documents have been written for IBM Sametime 8.5.2 IFR1, but they are still very valuable for S9.

But, the Audio/Video media framework has slightly changed, so I would like to share with you additional elements. IBM Sametime 9 now includes a "true" MCU, providing continuous presence and the support of SIP SVC (and AVC).

In a Sametime Meeting, you can do a dial out to an endpoints via moderator Control (picture 1), but now you can also dial in. You can retrieve dial in information from conference details (picture 2).

Dial-out

Dial-in

IBM Sametime 9 & SIP trunk architecture

The four main component of the A/V media framework are :

• SIP Proxy/registrar (P/R)
• Conference Focus
• VMgr
• VMcu 

These components are linked together via REST API or SIP trunk. I have tried to summarize it in the below schema.

In order to link Sametime 9 with the outside world, I have added two SIP trunks (black lines):

• Between external SIP server and SIP P/R
• Between SIP P/R and VMgr

My "simple" dial plan:

• S9 VMgr meeting: 10xx
• SIP Endpoints: 12xx

That is my rules (order is important)

• Rule N°1 (dial-in) redirects incoming calls for the Sametime Video meeting to the VMgr (then Vmcu)

That means, calls coming from 10.47.49.164 that have 10xx as pattern should be routed to the VMGR (FQDN: stvideomgr.ucalab.polycom.com). A dial-in flow like this doesn't involve the conference focus.

• Rule N°2 redirects incoming calls to Sametime Client.

That means, calls coming from 10.47.49.164  should be routed to the Conference Focus (FQDN: sametime9.ucalab.polycom.com). A dial-in flow like this involves the conference focus.

• Rule N°3 (dial-out) redirects calls to external SIP server

That means, calls coming with 12xx as pattern should be routed to the external SIP proxy (@: 10.47.49.164).

Hope it will help to deploy SIP trunks with IBM Sametime 9.

Now, IBM Sametime 9 Meeting with content sharing and Audio/Video

Today, I have finalized IBM Sametime 9 Meeting deployment. The deployment of Meeting & Audio/Video services is really simple and straightforward.

 

Now, I'm waiting for the Sametime Meeting mobile app.

My first experience with IBM Sametime 9 audio & video service

Today, I have just finished the deployment of my IBM Sametime 9 platform with the new SVC media framework.And, I have done tests with colleagues of point to point calls and multipoint calls with continuous presence on the new IBM Video MCU.

Point to Point call: (works with all Sametime bundles)

Multipoint Point Calls: (required IBM Sametime Complete)

They have also refreshed the "Audio & Video Performance" view.
You have now a better view on th audio/video traffic (Bit Rate, Resolution, Jitter...).

More tests to come, especially on Polycom interoperability.

IBM Sametime 9 is DOWNLOADABLE !!!

GREAT NEWS, you can now download and install IBM Sametime 9...

I know what I will do next week :) :) :)