OCS R2 Mediation Server Req RRS feed

  • Question

  • Hi all,

    we are installing R2 and mediation server alsong wiht IP GW to provide end users with external [national/international etc] calling at our Primary site A.  We would also like to enable another site [B] with OCS.  The site B doest not have any PBX.  The question is, do we need to put an IP GW in the second site B so that it can be connected to primary site's PBX?  Also, in order for site B end users to make Internal and Ext calls, is there a Mediation server required at site B? or can we just get by installing only a IP GW (audiocodes) connected to primay site's pbx?

    Thanks All for replies.

    Thursday, April 16, 2009 4:20 PM

All replies

  • OCS needs a 1:1 ratio of Mediation Servers per gateway/IP-PBX at each site with a phone system.  Since you only have a single phone system then users in remote sites would be configured to use the only mediation/pbx combination in your environment.
    Jeff Schertz, PointBridge | MVP | MCITP: Enterprise Messaging | MCTS: OCS
    Thursday, April 16, 2009 6:31 PM
  • It depends very much from the physical location of the two sites. This is - if they are in different Calling Rate Centers, you might endup with some charges.

    Consider this scenario:

    Site A - Seattle
    Site B - Atlanta

    Your Mediation/Audiocodes is deployed in Seattle. When a user from Atlanta dials ATLANTA LOCAL NUMBER, because the call will ordinate from Mediation deployed in Seattle, the call will be charged Lond Distance. Of source, when Atlanta dials number in Seattle, this will be a "local call" and not charged.

    Bottom line, if the sites are in different Rate Calling Centers, it would be benefitial to have two sets of Mediation/Gateways. And yet another tricky part - if instead of Gateways, you connect your Mediation server(s) to ITSP, then you could have two Mediation servers in Seattle, each one connected to a SIP trunk "terminating" for the appropriate Rate Center.

    Thursday, April 16, 2009 6:39 PM
  • Jeff -

    Thanks for the reply.
    So, if I have no PBX, IP GW and Mediation server at Site B,  the users can connect directly to the IP GW in Site A?  Or

    A better option would be to add a local IP GW  and Mediation server [connected to the ocs pool in site A] at site B and have users connect their SIP phones to Site B IP GW?

    Also, if the OCS servers are down in the primary Site A, then users at Site B wont be able to make any calls? Correct?
    Thursday, April 16, 2009 6:58 PM
  • Drago,

    THx for the reply.

    So, the answer you are suggesting is put another set of IP / GW and Mediation server at Site B? Correct?  But this may cost long distance charges.  But cant the users make SIP calls from site A to SIte B without getting charge anything?

    I understand, that if the users in Site B make a Local, national etc external call, the call with route out to IP GW [site B] -> PBX [Site A] -> Telco PSTN; which will incurr charge from the Telco.  Correct?

    Thursday, April 16, 2009 7:02 PM
  • First and mandatory condition - the two sites must be connected with VPN. Typically this is achieved with T1 lines which are very reliable. If you want local survivability, another mandatory condition is to have (writable) domain controllers in each site.

    Now, if you deploy OCS and Mediation in each site, you will achieve very high level of survivability, because even if Site A is down or unreachable, Site B will be Voice capable.
    When all conditions are true, this is what happens:

    Call from Site A (domain user) to Site B (domain user) and vice versa is free because never leaves your organization.

    Call from Site A to number in Site’s B Call Rate Center*** and vice versa (when the routing is done properly) is free.

    ***For example, My Rate Center is Milledgeville, GA. All calls to NPA 478 and NXX - 223 251 288 295 363 387 414 445 451 452 453 454 456 457 491 696 776 804 are considered “local”. So, when a user form Atlanta dials 478-445-2705, the call is placed via Mediation server in Milledgeville (handles the Milledgeville local Rate Center) and the call is free for the Atlanta user.

    Thursday, April 16, 2009 7:27 PM
  • Drago,

    Thx for the reply.

    I understand that logic.  I know there a requirement for 1 mediation serve : 1 IP GW.  So if I do install a mediation at Site B, i will certainly need a IP GW.  Now, what i just install the IPGW in site B and tie it to the PBX from SIte A?  When a user in site B dials even a local call, that could route to site A through the site B IP GW, which could i incur costs. Correct. 

    Also, is there a requirement that user objects in AD must have their 'telephoneNumber' attribute filled with correct #? otherwise the IP GW / UM / OCS FE wont be able to route call?

    Friday, April 17, 2009 5:44 PM
  • The Line URI field (in the user's Communications tab under Telephony options) is what must be populated for correct call routing within OCS.  But it's best practice to also populate the Work Phone field with the same number (typicall in E.164 format but not required) so that click-to-dial calling and reverse name lookups will work correctly as the Address Book service pulls phone numbers from the standard AD phone attributes (telephoneNumber, otherPhone, etc), not the OCS-specific Line URI (msRTCSIP-Line).
    Jeff Schertz, PointBridge | MVP | MCITP: Enterprise Messaging | MCTS: OCS
    Friday, April 17, 2009 6:07 PM
  • Jason,

    First of all, you need to decide (after research), what type of termination to use.

    If you want to use your existing phone lines, this is - to plug it in to the Gateway of type AudioCodes Mediant 1000 instead of your phones, and take advantage of UC, that's OK. In this case, your Gateway MUST be located in the site where it belongs. Indeed, you can route to the Mediation -> OCS located in Site A (your main office) for example. Same for T1 (PRI) trink from your local carrier.

    If you want to use SIP Trunk, you can connect the trunk to a Mediation server on the Moon if there is Internet connection. Some say there is...(smile).

    I can tell you right away that SIP Trunking is more easy, cost and growth effective.


    Friday, April 17, 2009 8:58 PM
  • Jeff,

    Thx for the reply.

    I am aware of the UC AD requrirements when it comes to user attrib. I meant to specifically ask this in terms of Exchange 2007 UM server.  when a call gets handed off the to the IP GW, it forwards it to the configured 'VOIP end points' i.e. the UM server.  At this point, how does the UM server know that e.g. +13056672233 belongs to John Doe? Does the UM server query the user obj/AD for 'telephoneNumber' attrib? 

    Sunday, April 19, 2009 4:58 PM
  • Drago,

    The secondary site will have an IP GW with T1 connection from the Site A PBX [QSIG/CAS]. The users will be equipped with either SIP USB phones and/or Nortel Tanjay [IP]. There may be some instance when end users may end up having Digital phone stations [Non IP] so that if OCS is down or primary site is down, then user can make external call.  RCC will be used for simultaneous call notification on all user end points.

    Thx for the reply.

    Sunday, April 19, 2009 5:08 PM
  • I am confused :-). Without having your full layout, I might confuse you as well. Get with me here: dragomir at gmc.cc.ga.us via email or federation if you have Edge server deployed do discuss your situation.

    Sunday, April 19, 2009 7:26 PM