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    Thanks again, Daniel, that is very helpful.<br>
    <br>
    BTW, I also found that even without disabling rtpproxy, most data
    would be transfered directly between endpoints.<br>
    Following is the bandwidth test result of video call between two
    Jitsi clients.<br>
    <br>
    <img alt="rtpproxy enabled"
      src="cid:part1.02050405.04010501@gmail.com" height="284"
      width="334"><br>
      <br>
       Figure 1. use_rtpproxy set as "Always with plain SDP"<br>
    <br>
    <br>
    <br>
    <img alt="rtpproxty disabled"
      src="cid:part2.03070206.03040805@gmail.com" height="281"
      width="354"><br>
    <br>
       Figure 2. use_rtpproxy set as "Never"<br>
    <br>
    <br>
    <div class="moz-cite-prefix">On 2014年10月27日 01:41, Daniel Grotti
      wrote:<br>
    </div>
    <blockquote cite="mid:20141026174152.3C5E6649B3@mail.sipwise.com"
      type="cite">
      <p dir="ltr">Hi,<br>
        Yes disabling  rtpproxy rtp will go directly from endpoint to
        endpoint, if they can see each others. If you have endpoints
        behind nat you need to use rtpproxy.</p>
      <p dir="ltr">To disable it you just need to do it via web
        interface, on domain/peer level under preferences you can find
        the parameter use_rtpproxy. Just set it to 'never'.</p>
      <p dir="ltr">But be careful if you have clients behind nat.</p>
      <p dir="ltr">Daniel<br>
      </p>
      <div class="quote">On 26 Oct 2014 16:59, yanzs
        <a class="moz-txt-link-rfc2396E" href="mailto:danrenjian@gmail.com"><danrenjian@gmail.com></a> wrote:<br type="attribution">
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          thanks, Daniel<br>
          <br>
          I spend a few days studying concepts like RTP, SIP and VOIP. <br>
          <br>
          So if I disable 'rtpproxy', RTP stream can be sent directly
          between endpoints (or at least without any intermediate server
          I owned), it that correct? I am concerned with that because
          many concurrent video calls would require a high bandwidth on
          the server, and that is a bit expensive.<br>
          <br>
          <b>And another question</b>, in the "Platform Architecture"
          chapter of SPCE handbook, it says " The mediaproxy
          configuration file is
          /etc/ngcp-config/templates/etc/default/ngcp-mediaproxy-ng-daemon"
          which I believe is where I can disable rtpproxy. But I could
          not find that "ngcp-mediaproxy-ng-daemon" path with spce 3.4.2
          installed by virtual box image, do I need to install spce
          using NGCP install CD to get this ngcp-mediaproxy-ng-daemon
          service ?<br>
          <br>
          Thanks in advance.<br>
          Russell<br>
          <br>
          <div class="moz-cite-prefix">On 2014年10月22日 16:10, Daniel
            Grotti wrote:<br>
          </div>
          <blockquote cite="mid:54476665.2090900@sipwise.com"
            type="cite">
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            <div class="moz-cite-prefix">Hi,<br>
              RTP stream (audio/video) just pass through NGCP, which is
              just an RTP relay.<br>
              You can even bypass NGCP during RTP session, by disable
              'rtpproxy' if you want. <br>
              It depends what do you need, if you need to perform some
              actions on rtp stream (rtptimeout for example), you need
              NGCP staying in the middle.<br>
              <br>
              Daniel<br>
              <br>
              <br>
              <br>
              <br>
              On 10/22/2014 09:38 AM, yanzs wrote:<br>
            </div>
            <blockquote cite="mid:54475EE8.3090301@gmail.com"
              type="cite">
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              Hi, community<br>
              <br>
              Our team are building  an application for users to make
              video call to each other (just for confirming their face
              and do some stuff in the system), and we are new to spce
              and even voip.<br>
              In the sipwise site, it claims that with these recommended
              hardware: <br>
              <br>
              <meta http-equiv="content-type" content="text/html;
                charset=utf-8">
              <ul style="margin: 0px 0px 10px 0.4em; padding: 0px;
                border: 0px; outline: 0px; font-size: 11px;
                vertical-align: baseline; list-style: disc; color:
                rgb(51, 51, 51); font-family: Arial; font-style: normal;
                font-variant: normal; font-weight: normal;
                letter-spacing: normal; line-height: 20.7999992370605px;
                orphans: auto; text-align: left; text-indent: 0px;
                text-transform: none; white-space: normal; widows: auto;
                word-spacing: 0px; -webkit-text-stroke-width: 0px;
                background: rgb(246, 246, 246);">
                <li style="margin: 0px; padding: 0px; border: 0px;
                  outline: 0px; font-size: 11px; vertical-align:
                  baseline; list-style: disc inside none; display:
                  block; background: transparent;">Dual-core x86_64
                  compatible</li>
                <li style="margin: 0px; padding: 0px; border: 0px;
                  outline: 0px; font-size: 11px; vertical-align:
                  baseline; list-style: disc inside none; display:
                  block; background: transparent;">3GHz, 4GB RAM, 128GB
                  HDD</li>
              </ul>
              <p>Sipwise can sustain <span style="color: rgb(51, 51,
                  51); font-family: Arial; font-size: 11px; font-style:
                  normal; font-variant: normal; font-weight: normal;
                  letter-spacing: normal; line-height:
                  20.7999992370605px; orphans: auto; text-align: left;
                  text-indent: 0px; text-transform: none; white-space:
                  normal; widows: auto; word-spacing: 0px;
                  -webkit-text-stroke-width: 0px; display: inline
                  !important; float: none; background-color: rgb(246,
                  246, 246);">2.000 Concurrent Calls, </span><span
                  style="color: rgb(51, 51, 51); font-family: Arial;
                  font-size: 11px; font-style: normal; font-variant:
                  normal; font-weight: normal; letter-spacing: normal;
                  line-height: 20.7999992370605px; orphans: auto;
                  text-align: left; text-indent: 0px; text-transform:
                  none; white-space: normal; widows: auto; word-spacing:
                  0px; -webkit-text-stroke-width: 0px; display: inline
                  !important; float: none; background-color: rgb(246,
                  246, 246);">50 Call Attempts per Second.</span><br>
                So I am wondering how audio and video stream data are
                transfered from one client to another, does clients make
                a p2p connection, or all data are passed through the
                spce server?<br>
              </p>
              <p>Thanks, <br>
                Russell<br>
                <span style="color: rgb(51, 51, 51); font-family: Arial;
                  font-size: 11px; font-style: normal; font-variant:
                  normal; font-weight: normal; letter-spacing: normal;
                  line-height: 20.7999992370605px; orphans: auto;
                  text-align: left; text-indent: 0px; text-transform:
                  none; white-space: normal; widows: auto; word-spacing:
                  0px; -webkit-text-stroke-width: 0px; display: inline
                  !important; float: none; background-color: rgb(246,
                  246, 246);"></span></p>
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