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Use plain sip phone directly on OCS R2? (not through IP PBX) RRS feed

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    Is it possible?

    I thought OCS FE doesn't understand G.711 (only Mediation), yet I have already integrated a SIP conferencing system that does use G.711 for voice and it DOES work!

    Wednesday, August 5, 2009 9:05 AM

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    AFAIK this isn't possible.  To connect to OCS you have to authenticate as a user, this requires an OCPE client based phone.  There are phones by SNOM out there that don't use OCPE, but they have a special software build on them that is made specifically for integrating with OCS.  Just a standard VoIP phone probably wouldn't be possible.

    -kp

    Kevin Peters MCSE/MCSA/MCTS/CCNA/Security+ blog: www.ocsguy.com
    Wednesday, August 5, 2009 11:43 AM
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    OCS and OC are not limited to just RT audio codec and siren (conferencing codec). The servers and client will all accept G711 and a bunch of other codecs. In fact I have heard of one deployment that only allows G711 to be used and no RT audio codec.

    The limiting factor is as Kevin states, you need to be able to authenticate to OCS and do TLS if you have this as a requirement. I would imagine this could be possible with some SIP phones but not supported and more than likely may require setting OCS to security supported but not required communication if the phone can not do TLS.

    Cheers
    Chris
    http://voipnorm.blogspot.com/
    Wednesday, August 5, 2009 2:19 PM

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    AFAIK this isn't possible.  To connect to OCS you have to authenticate as a user, this requires an OCPE client based phone.  There are phones by SNOM out there that don't use OCPE, but they have a special software build on them that is made specifically for integrating with OCS.  Just a standard VoIP phone probably wouldn't be possible.

    -kp

    Kevin Peters MCSE/MCSA/MCTS/CCNA/Security+ blog: www.ocsguy.com
    Wednesday, August 5, 2009 11:43 AM
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    OCS and OC are not limited to just RT audio codec and siren (conferencing codec). The servers and client will all accept G711 and a bunch of other codecs. In fact I have heard of one deployment that only allows G711 to be used and no RT audio codec.

    The limiting factor is as Kevin states, you need to be able to authenticate to OCS and do TLS if you have this as a requirement. I would imagine this could be possible with some SIP phones but not supported and more than likely may require setting OCS to security supported but not required communication if the phone can not do TLS.

    Cheers
    Chris
    http://voipnorm.blogspot.com/
    Wednesday, August 5, 2009 2:19 PM
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    well if the limiting factor is TLS then it is a non-issue

    1) you can treat specific addresses as authenticated
    2) you can add 5060/TCP under 5061/TLS in front-end settings
    3) you can configure the endpoint to use TLS anyway

    as I said I already have an endpoint (a conference system non-OCS related/based) that uses G711 for voice and it DOES work (both as TCP and as TLS)



    Sunday, August 23, 2009 6:40 AM
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    Hi,
    The story is that you can't register a non-OCS endpoint to OCS. So you can't take a plain old Polycom 550 SIP phone and have it register as an endpoint of OCS.

    However - you can have certain other conferencing systems like PolyCom or Tandberg bridges (like RMX) register on behalf of non-OCS devices. So the non-OCS devices talk to the conferencing system, then the conferencing system talks to OCS FE server. It's definitely supported & works.

    Both PolyCom and Tandberg also make endpoint devices like HDX series that can register to OCS using an AD username and password. But these really couldn't be considered "plain SIP phones".

    And you are correct, of course, about TLS / TCP. Both are supported by OCS.


    Regards,
    Matt

    Matt McGillen, PointBridge - https://blogs.pointbridge.com/Blogs/mcgillen_matt/default.aspx
    Tuesday, August 25, 2009 7:15 PM