<br><br><div class="gmail_quote">On Fri, Mar 16, 2012 at 12:37 PM, Marcelo Pacheco <span dir="ltr">&lt;<a href="mailto:marcelo@m2j.com.br">marcelo@m2j.com.br</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

  
    
  
  <div bgcolor="#FFFFFF" text="#000000">
    Gustavo,<br>
    <br>
    I think you&#39;re confusing the general function of an STP with the
    external signaling network architecture used by ANSI countries.<br>
    <br>
    All incumbent networks in Brazil make heavy usage of STPs.<br>
    They have lots of </div></blockquote><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div bgcolor="#FFFFFF" text="#000000"> TDM switches, and to avoid a full mesh of
    signaling links between all TDM switches that have voice trunks
    between them, STPs are used to aggregate SS7 traffic.<br></div></blockquote><div><br></div><div>STP is single point of failure unless used in pairs; using STP pairs requires combined linkset - does ITU have this capability?  don&#39;t think so - SLS is only 4 bits; it&#39;s 5 bits in ANSI </div>
<div><br></div><div>of course it is what it is - but i&#39;m curious how the fault tolerance vs management ease balance out - mildly curious</div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
    <br>
    Also STPs are also used as billing entities and for resolving LNP in
    some carriers.<br>
    <br></div></blockquote><div>This seems to me to be the motivation for using STPs.</div><div> </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div bgcolor="#FFFFFF" text="#000000">

    I&#39;m pretty sure STPs have lots of usage in other ITU countries.<br>
    <br></div></blockquote><div>Poland. &quot;Lots&quot; is a relative term.  You do see them.  Seems like they were being introduced about the same time that VoIP was coming in too.  Now what? Keep building TDM or cap it and go to VoIP?  I think we all know how that is turning out.</div>
<div> </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div bgcolor="#FFFFFF" text="#000000">
    However they don&#39;t have a fully separate signaling network, 64kbps
    SS7 links make maximum usage of semi permanent call setups,
    specially for interconnects with other carriers (using bearer
    channels of existing E1 voice trunks).<br>
    <br>
    However competitive carriers use redundant soft switch architecture
    don&#39;t need STPs, since signaling flows through the IP network,
    without explicit signaling channels.<br>
    <br>
    I fell more important than the capability of Asterisk performing as
    an STP, is much more important full linkset functionality as a
    regular signaling point. For instance, the following scenario can&#39;t
    be implemented with libss7 today:<br>
    <br>
    <tt>Asterisk --x-- STP A ---x--- Switch1,2,3,4,5,6,7,8<br>
                     STP B<br>
    </tt><br>
    Where Asterisk has voice CICs with all 8 switches, and all signaling
    needs to be shared across a pair of signaling links, one with each
    STP. Specially with E1s with all 8 switches can&#39;t fit on a single
    Asterisk box.</div></blockquote><div><br></div><div>Are you describing the combined linkset?  When I&#39;ve seen things like this in ITU networks, A was primary and B was alternate (used when A was not available), instead of the ANSI model where A and B are peers and normally used equally using a 5 bit SLS.</div>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div bgcolor="#FFFFFF" text="#000000"><span class="HOEnZb"><font color="#888888"><br>
    <br>
    Marcelo Pacheco</font></span><div><div class="h5"><br>
    <br>
    On 03/15/12 14:39, Gustavo Mársico wrote:
    <blockquote type="cite"><br>
      <div>
        <div>On Mar 15, 2012, at 2:17 PM, Michael Mueller wrote:</div>
        <br>
        <blockquote type="cite">STPs in ITU-land are awkward since ITU
          voice networks are a mesh of E1 with signaling in the same
          bundles as the voice
          <div> <br>
            <div>in ANSI-land, the STP was incorporated and mandated by
              two large and powerful monopolies: BC and ATT; signaling
              became de-coupled from the voice and traveling in a
              separate network connected by hierarchy of mated pair STPs</div>
            <div><br>
            </div>
            <div>putting an STP or an STP-like invention in an typical
              ITU network raises questions about commercial viability:
              having a central STP might raise your E1 charges because
              they travel over longer distances - this raises monthly
              charges in many places; might be cheaper to connect
              locally - but then you have increased monthly charges for
              colo space</div>
            <div><br>
            </div>
            <div>there is conceptual dissonance between STPs and ITU
              networks - STPs require the signaling be separate from the
              voice, and ITU mesh networks are built around signaling
              and voice channels traveling in the same bundles of wire
              (i&#39;ve just restated my first 2 paragraphs); decoupling
              signaling means using an entire E1 for a single signal
              channel; this usually causes despair to the typical ITU
              ss7 engineer but is business as usual to the ANSI
              counterpart</div>
          </div>
        </blockquote>
        This is not quite correct. CALA region mostly uses separate E1
        for signalling and media when a STP is used. If STP is not
        required, some telco choose to separate and others don&#39;t. As the
        same as ANSI does. In fact, it&#39;s a matter of how the people
        wants to make it work.</div>
      <div><br>
        <br>
        <blockquote type="cite">
          <div>
            <div>the cheapest STP I know of is the PT Segway; maybe you
              can get a Tekelec Eagle; I&#39;m not aware of any Linux based
              DIY STPs; ss7box started as an STP but evolved away from
              the function as there was little need for a low-end STP in
              ANSI-land and zero need for it in ITU-land</div>
            <div><br>
            </div>
            <div>ss7box supports Asterisk box clustering around a single
              point code with CIC routing; clustering might be something
              you want to investigate - you&#39;d have to examine the
              technical, commercial, and incumbent connection policies
              to see if it would help you build an IP voice network with
              fewer connections to the incumbent telco network using
              such a clustering function </div>
            <div><br>
            </div>
            <div>you asked a complicated question, or I&#39;ve turned a
              simple question into a complicated one - both are
              plausible</div>
            <div><br>
              <div class="gmail_quote">On Thu, Mar 15, 2012 at 11:45 AM,
                Rodrigo Ricardo Passos <span dir="ltr">&lt;<a href="mailto:rodrigopassos@gmail.com" target="_blank">rodrigopassos@gmail.com</a>&gt;</span>
                wrote:<br>
                <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
                  <div bgcolor="#FFFFFF" text="#000000"> Michael,<br>
                    <br>
                    Can you explain more?<br>
                    Here, in Brazil, the standard is ITU. I think it
                    isn´t possible because ITU is used in all telcos.<br>
                    <br>
                    <br>
                    Em 15/03/2012 12:25, Michael Mueller escreveu:
                    <div>
                      <div>
                        <blockquote type="cite">connecting a mated pair
                          of STPs to an ANSI network as a peer has more
                          requirements than connecting an SSP; ITU STP
                          are less common so connection requirements
                          might be more variable<br>
                          <br>
                          <div class="gmail_quote">On Thu, Mar 15, 2012
                            at 10:19 AM, Rodrigo Ricardo Passos <span dir="ltr">&lt;<a href="mailto:rodrigopassos@gmail.com" target="_blank">rodrigopassos@gmail.com</a>&gt;</span>
                            wrote:<br>
                            <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
                              <div bgcolor="#FFFFFF" text="#000000">
                                Gustavo,<br>
                                <br>
                                Do you know Yate? Knows if Yate can be
                                used in place of asterisk?<br>
                                I know that this list is about Asterisk
                                SS7, but I think that this question
                                doesn´t bad.<br>
                                <br>
                                Regards,<br>
                                <br>
                                Rodrigo<br>
                                <br>
                                Em 14/03/2012 16:55, Gustavo Mársico
                                escreveu:
                                <blockquote type="cite">
                                  <div>There is no pure STP
                                    implementation on libss7 or
                                    chan_ss7. Modules cannot send TFA,
                                    TFP, support STP timers, etc. Today,
                                    all you can do is routing based in
                                    the extensions, but that&#39;s not STP
                                    function.</div>
                                  <div>However, I know that some efforts
                                    were made on libss7 and the last
                                    time I checked looked promising.
                                    I&#39;ll try to find what&#39;s going on
                                    there.</div>
                                  <div><br>
                                  </div>
                                  <div>Regards</div>
                                  <div><br>
                                  </div>
                                  <div>Gustavo</div>
                                  <div><br>
                                  </div>
                                  <br>
                                  <div>
                                    <div>On Mar 14, 2012, at 4:39 PM,
                                      Rodrigo Ricardo Passos wrote:</div>
                                    <br>
                                    <blockquote type="cite">
                                      <div bgcolor="#FFFFFF" text="#000000"> Hi all,<br>
                                        <br>
                                        <p class="MsoNormal"><span lang="EN-US">I have a
                                            question of how can I create
                                            STPs Boxes with Asterisk in
                                            my network?</span></p>
                                        <p class="MsoNormal"><span lang="EN-US">My project
                                            includes a creation of
                                            network with asterisk SPs
                                            and STPs and my initial idea
                                            is a implementation of these
                                            boxes using TDMoE. So,
                                            create two boxes like STP e
                                            another’s boxes like SP.</span></p>
                                        <p class="MsoNormal"><span lang="EN-US">All
                                            signalization will pass to
                                            both STPs. Anyone knows if
                                            my scenario will be one
                                            scenario with a real STP
                                            boxes or this will never STP
                                            ambient?</span></p>
                                        <p class="MsoNormal"><span lang="EN-US">Other question
                                            is, if this last question is
                                            false, how can I create this
                                            ambient with asterisk?<br>
                                          </span></p>
                                        <p class="MsoNormal"><br>
                                          Best Regards,<br>
                                        </p>
                                        <p class="MsoNormal">Rodrigo<br>
                                          <span lang="EN-US"></span></p>
                                        <br>
                                        <span><font color="#888888"> </font></span></div>
                                      <span><font color="#888888"> --<br>
_____________________________________________________________________<br>
                                          -- Bandwidth and Colocation
                                          Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a>
                                          --<br>
                                          <br>
                                          asterisk-ss7 mailing list<br>
                                          To UNSUBSCRIBE or update
                                          options visit:<br>
                                            <a href="http://lists.digium.com/mailman/listinfo/asterisk-ss7" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-ss7</a></font></span></blockquote>
                                    <span><font color="#888888"> </font></span></div>
                                  <span><font color="#888888"> <br>
                                      <br>
                                      <fieldset></fieldset>
                                      <br>
                                      <pre>--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a> --

asterisk-ss7 mailing list
To UNSUBSCRIBE or update options visit:
   <a href="http://lists.digium.com/mailman/listinfo/asterisk-ss7" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-ss7</a></pre>
                                    </font></span></blockquote>
                              </div>
                              <br>
                              --<br>
_____________________________________________________________________<br>
                              -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a>
                              --<br>
                              <br>
                              asterisk-ss7 mailing list<br>
                              To UNSUBSCRIBE or update options visit:<br>
                                <a href="http://lists.digium.com/mailman/listinfo/asterisk-ss7" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-ss7</a><br>
                            </blockquote>
                          </div>
                          <br>
                          <br>
                          <fieldset></fieldset>
                          <br>
                          <pre>--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a> --

asterisk-ss7 mailing list
To UNSUBSCRIBE or update options visit:
   <a href="http://lists.digium.com/mailman/listinfo/asterisk-ss7" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-ss7</a></pre>
                        </blockquote>
                      </div>
                    </div>
                  </div>
                  <br>
                  --<br>
_____________________________________________________________________<br>
                  -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a>
                  --<br>
                  <br>
                  asterisk-ss7 mailing list<br>
                  To UNSUBSCRIBE or update options visit:<br>
                    <a href="http://lists.digium.com/mailman/listinfo/asterisk-ss7" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-ss7</a><br>
                </blockquote>
              </div>
              <br>
            </div>
          </div>
          --<br>
_____________________________________________________________________<br>
          -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a>
          --<br>
          <br>
          asterisk-ss7 mailing list<br>
          To UNSUBSCRIBE or update options visit:<br>
            <a href="http://lists.digium.com/mailman/listinfo/asterisk-ss7" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-ss7</a></blockquote>
      </div>
      <br>
      <br>
      <fieldset></fieldset>
      <br>
      <pre>--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --

asterisk-ss7 mailing list
To UNSUBSCRIBE or update options visit:
   <a href="http://lists.digium.com/mailman/listinfo/asterisk-ss7" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-ss7</a></pre>
    </blockquote>
    <br>
  </div></div></div>

<br>--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
<br>
asterisk-ss7 mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
   <a href="http://lists.digium.com/mailman/listinfo/asterisk-ss7" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-ss7</a><br></blockquote></div><br>