Questions on POD

Comments

Question: Where do I get the PDF files - presentations and lab documents?

Answer: Using your openVPN access data you can find these documents on our SAMBA server. The instructions and the IP address is provided in the document "SUT_Prep_Client_Installation.pdf". This document is attached to email that you received after your registration is accepted for the course.

Questions on SUT Integration / Implementation

Information: SUT8 CF1 is the first release that Advanced and SUT are tested together by SUT FVT team.

So only the two combinations we tested so far:

1. Standalone: Sametime8.0.2 + Sametime8.0.2 hotfix + SUT8.0 CF1 + Advanced8.0.1 CF1

(Note that 8.0.2 hotfix is the dependancy for installing SUT8.0 CF1 on windows platform, on other platforms you can ignore it, the sut/adv install sequence doesn't matter, you may install SUT before Adv))

2. Notes Embedded: Notes8.5.1(+FP1) + Advanced8.0.1 CF1 for Notes8.5.1 + SUT8.0 CF1 (the sut/adv install sequence doesn't matter, you may install SUT before Adv)

It doesnot matter you install SUT or Advanced first.

Question: The status for SRTP encryption seems to be : "The SUT soft phone to SUT soft phone can be encrypted. But, in the current release, encrypted communication are not yet supported between soft phone to desk phone." But where SRTP is to be set to encrypt soft phone-soft phone ? from CMP (we see only CMP settings for SRTP media-server) ? from client side ? and why it does not work from soft phone to desk phone ?

Answer: SRTP is not yet supported, but it should work in the 8.5.1 stream against the Siemens media server and gateways. It has been tested against at least one third party softphone in 8.5.1.

Question: Origin depending routing in conference (i.e. use the MS that is reside at that particular site - MS in US for the subscribers in US and MS in Germany for subscribers in DE). We are in SUT-pilot project. Do we have this scenario - say MS in Ireland for you, MS in US for US subscribers and MS in DE for subscribers in Germany?
If yes, who has configured this?
If no, who can give more information about the such type of configuration?
Is there any documents that we can provide to our SUT students who are from IBM?

Answer: This is an interesting question and one that has no simple answer. Can we support multiple MS's in multiple countries? yes is the simple answer. But there is a one to one relationship between TAS's and MS's. And as you know you have an upper limit of 8 TAS's. So having an MS per country is not going to work. However that should not effect localization concerns, if that was one of your concerns.

The balance of TAS's to countries should be done purely on performance concerns. If you have a remote department were there will be a lot of users but there exists a bad data trunk to the main offices, then a TAS/MS pair may be justified at that remote location to reduce RTP traffic. Maybe an American company with offices in China, Japan and Korea would have one TAS/MS there. It very much depends on projected usage (particularly on conferecing).

In Dublin we have 3 TAS's hanging off the one TCS, all operating with different country codes. For the sanity of the developers using these machines, we didn't change the lanaguages on them, so if your questions relates to localization its not implemented in Dublin, but its straightforward.

The documentation for intermediate talks about having multiple NP's for different geographies, and the use of a CNP. In such a deployment, all TAS's would likely reside in the CNP so conferencing can be supported across all the geographies with the benefit of Least cost routing

Question: "The first one is an architecture topic: I would like to know what needs to be set up to be able to have the following feature: I have one TCS and 2 pbx connected to TCS, the first one defined as primary SIP trunk​. Can we set up a secondary route within TCS with a different least cost routing in order to pass the sig to the secondary pbx if the first one is down ? "

Answer: Yes, we can configure TCS the way that it has a primary PBX1 for the normal case and a redundant PBX2 for the case that PBX1 is not available. The element that hold this configuratin is the "destination". A destination could contain several routes to several endpoints (PBXes). The algorithm can be configured if the behaviour is managed with priorities or in a round robin manner. But the routes in a destination are not usefull for Least cost routing.

Least cost routing is configured in the routing configuration in the PAC and destination code tables. Least cost routing analyses a number and chooses the destination that would match to the "least cost criteria". This decision is a static decision. Example1: We have a contract with an operator for all cheap international calls to Japan. The operator is reachable via the PSTN by a special prefix. The routing configuration needs an entry for all international calls to Japan, the operator prefix must be added and the call goes to the PSTN.

Finally to make the configuration perfext you could combine "least cost routing" and "redundant routes" and use for the leased cost routing a destination that has a primary route to PBX1 and secondary route to PBX2.

Answer: The key important thing you are asking about "leverage SUT conference bridge" is not in the initial SUT release and unfortunately is still not in plan for 8.5.1. We discussed having it in the initial version of the product, but there was no good way to prevent non-SUT users from using SUT services. As you may be aware the conference bridge in SUT does not scale-up separately from the TAS--there is at best a single stand-alone media server per TAS so there really is no good way to open up conference services as a separate usable feature beyond the click-2-conference/adhoc conferencing we do from the SUT client. In 8.5.1, we could have used policy controls to allow only SUT users to have this capability, but 8.5.1 was a minimum feature release, so we had no time to do something like this. Look for better integration with SUT and meetings/web conferences in 8.5.2.

Question: From theST 8.5 meetings UI, will it be possible for each attendee to indivually C2C a selected contact who is attending the meeting and see his phone presence ?

Answer: In 8.5.1, yes it should be possible. In 8.5.2, with better integration between SUT and meetings, there may be different presence and call options on the user names appearing in a meeting. For example, if we are all on the same conference call, seeing an off-hook probably doesn't help us much. It might be more useful to only show a hook state if one of the users makes or answers a second call so that the other people in the phone might guess he is not listening to the conference at the moment.

Question: Will it be possible to enrich the ST 8.5 meetings with third-party audio-bridges the same way it was done under 8.0.2 ?

Answer: Yes. It is possible to add 3rd party service providers to 8.5 and 8.5.1 that can be used in a meeting / web conference.

Question: Will it be possible to enrich the ST 8.5 meetings with the S.U.T. MS Conference bridge?

Answer: No. Unfortunately, this integration is not in plan for 8.5.1. Look for better integration in 8.5.2

Question: How strong does the machine hosting a TAS/TCS VMWare POC deployment need to be?

Answer: A POC of up to 100 users can be run using a virtual machine for both the TAS and TCS. It is recommended that each of the VMs have at least 4 GB available RAM and 2 processors. Given this, if you want to run both VMs on the same VMWare host, you should have at least a 10 GB physical machine with at least 4 processors.