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    Gateway/ Gatekeeper for VoIP/ H323

    salam techies!

    direct to the point,
    i'm working on a project in which we have to initiate a VoIP/ H323 call to a PSTNetwork using some VoIP Gateway/ Gatekeeper, i'm facing some problems like when we initiate call, the receiver (the person who's on the phone) can hear my voice, but when he speaks i cant hear his/her in my H323 client, to test this protocol right now we are using the standard implementation(Openphone) of OpenH323.org

    1) any ideas regarding what the problem could be? is there anything wrong with the proxy server (ISA)

    and yes, are behind a router, we can access the router directly and/or from behind a proxy both situations are possible, i've not tested it on dialup connection, it may work there but i want this solution for LAN

    2) second thing is that, is there any free VoIP gateway/Gatekeeper avaialble using which we can easily terminate our call to the local PABX ???? hoga to zaroor shaid, if anybody can point out the exact location on the web from where i can find out such software, it will be very helpfull

    thnx in advance

    raheel
    Quwwat-e-Mazhab say mustehkam hay jami'yat teri

    #2
    heh....I was looking at 100% the same thing today...I mean really...100% the same thing. spooky!

    Anyway, pc-telephone is one, which should let you make analog calls to PSTN through an ISDN line. They say it'll support TAPI3.0 soon for PSTN support.

    I started playing with OpenH323 a couple of days ago...lets see what happens.

    Btw...do tell us how you have setup your things and what hardware are you using.

    Comment


      #3
      acha, itaffaq hay
      i'll keep posting u if there's any progress

      anybody else knows anything about the topic, which may help us?
      Quwwat-e-Mazhab say mustehkam hay jami'yat teri

      Comment


        #4
        ok who-me, here's the architecture/ network diagram i'm using

        [thumb=A]VoIP.JPG[/thumb]
        Quwwat-e-Mazhab say mustehkam hay jami'yat teri

        Comment


          #5
          You might want to try RTP and TCP header compression and see if that helps. If you are already using TCP and and RTP compression, take it off the interface and apply the compression at the same time on both ends. It would be helpful if we can see your router configs. This only applies to Cisco Voice boxes.

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