Asterisk as a softswitch in a IP-Centrex environment?

Discussion in 'VOIP' started by Andre Chrcanovic, Apr 27, 2004.

  1. Hi all!

    We want to develop an IP-Centrex solution based on a softswitch
    architecture and want to hear from VoIP community their experiences,
    since Asterisk seems to be a good candidate for the job.

    Basics of the architecture are:

    * SIP IP Phones (who will be using the IP-Centrex services) connected
    through a managed IP network to the softswicth;
    * Softswitch connect through LAN to a signaling/media gateway, using a
    proprietary signaling/media gateway control protocol (no MGCP/H.248
    for now, unfortunately);
    * Signaling/media gateway connected to the PSTN using SS7 on the TDM
    side;
    * Signaling gateway is SIP capable on the IP side;
    * Media gateway is capable of: codecs (of course!), conferencing, DTMF
    tones detection/generation, message recording/playback using an
    external NFS server as a media server;

    For a nice diagram of this kind of architecture, please take a look
    at: http://www.ip-centrex.org/how/softswitch.html.

    Some numbers to give an idea of the "size" of the system:

    * Maximum number of IP-Centrex clients: 5,000;
    * Signaling/media gateway connection to the PSTN: DS3 (672 DS0's);
    * Signaling/media gateway DSP "resources": 1,024 (that can be
    configured for codecs, conferencing, DTMF tones detection/generation,
    message recording/playback);

    We want Asterisk to:

    1. If the caller and the called parties are both SIP clients, route
    the packetized voice streams directly to one another; consequently the
    voice stream never reaches the Asterisk box (or boxes);
    2. If the called party is served by the PSTN, instruct the originating
    IP Phone to route the packetized voice stream the media gateway;
    3. Based on (1.) above, use the media gateway DSPs conferencing
    capabilities in the conference feature;
    4. Based on (1.) above, use the media gateway DSPs message
    recording/playback capabilities in the voice-mail and IVR features;

    We understand the we may have to:

    * Develop of an Asterisk channel interface so the Asterisk can control
    the signaling/media gateway;
    * Adapt Asterisk conference, voice-mail, IRV to achieve (3.) and (4.)
    above;

    I'm looking forward to hear from the VoIP community their
    thoughts/comments on the Asterisk use in such a development:

    * How Asterisk adapts to such number of IP clients (5,000)?
    * Has Asterisk been used to control a signaling/media gateway in the
    way we are planning to do?
    * How hard it is for Asterisk to achieve requirements (1.) to (4.)
    above?

    Besides that, what are your experiences with Digium commercial
    Asterisk support, especially in a project with such size/requirements?

    Thanks in advance.

    André Chrcanovic
    FITec Inovações Tecnológicas

    Tel: +55 31 3263-4016
     
    Andre Chrcanovic, Apr 27, 2004
    #1
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  2. Andre Chrcanovic

    Soren Rathje Guest

    "Andre Chrcanovic" <> wrote in message
    news:...
    > Hi all!
    >
    > We want to develop an IP-Centrex solution based on a softswitch
    > architecture and want to hear from VoIP community their experiences,
    > since Asterisk seems to be a good candidate for the job.
    >


    Have thought of the same and came up with this..

    http://www.sentito.com/pdf/nsp.pdf

    plus Asterisk, unfortunately I never got a chance to pursue the matter..

    -- Soren
     
    Soren Rathje, Apr 27, 2004
    #2
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