How does SIP work?

Discussion in 'VOIP' started by Vincent Delporte, Sep 26, 2006.

  1. Hi

    I finally got to have a working set up using an Axon Windows PBX
    software, Linksys 3102 gateway, a GrandStream IP phone and an X-Ten
    softphone over the Net... but I don't know _why_ it works :)

    Here's how I think the whole thing works:

    1. I set up the router to map UDP 5060 to the host where the PBX is
    installed, and I launch the Axon server

    2. Remote phones connect through the Net into the Axon server to
    register their IP address and extension

    3. When a call comes in from the PSTN network into the Linksys, the
    3102 sends an SIP notification to the PBX. The PBX checks what
    extensions it must ring, and sends out SIP notifactions to all
    extensions involved. For this to work, all remote routers must also
    forward SIP messages to the IP phones that registered (UDP 5060 by
    defaullt, but each phone needs its own port to be reachable, eg. UDP
    5060 for the first phone in the LAN, UDP 5061 for the second phone,

    4. Once a phone goes off-hook, a connection is set up between the
    phone and the Linksys gateway. During the connection, each device
    tells the other what UDP ports it will use for RTP, ie. data packets.

    Provided this is correct so far, here's where things begin to blur:

    - If I don't set up remote phones to use STUN, connections are made,
    but I don't get sound in one direction: Is it because without STUN,
    the misconfigured phone sends its private IP in the data part of an
    SIP message, eg., and since this is an unroutable address
    the other device won't be able to route data packets?

    - I didn't forward any ports for RTP, but calls still work: Is it
    because I happen to have UPnP-capable routers, hence RTP ports are
    automagically opened to make things happen?

    Thanks much for any hint :)
    Vincent Delporte, Sep 26, 2006
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