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    <div class="moz-cite-prefix"><tt>Hi Maxim,<br>
        <br>
        It is good to know about the rtp_cluster, but aside simplifying
        things, it does not bring any new functionality - the LB and
        failover between RTPproxy nodes can be done now in OpenSIPS
        module .<br>
        The most challenging thing we are looking at is the ability to
        move calls between different instances of RTPP (for HA
        purposes)..or some restart persistence for the sessions -
        without something like that it's very hard to deal with SW/HW
        failures ; there are ways to go around for scheduled
        stops/restarts (maintenance), but non for unexpected failures. 
          <br>
        <br>
        Thanks and Regards,<br>
      </tt>
      <pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
      On 13.06.2014 00:36, Maxim Sobolev wrote:<br>
    </div>
    <blockquote
cite="mid:CAH7qZft=CmyO7nFMsjxCiHQFQBF8e=rEg5SWNRFHKVWEUmhmxw@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div>Brett, on the HA/carrier-grade side there is
          little-advertized middle-layer component called "rtp_cluster",
          which in essence is load-balancing, transparent dispatcher
          that can be inserted in between some call-controlling
          component like OpenSIPS or Sippy B2BUA and bunch of RTPP
          instances running on the same or multiple nodes. From the
          point of view of that OpenSIPS it's just another RTPP
          instance.<br>
          <br>
          And it handles all logic necessary to load-balance incoming
          requests between online instances plus it can handle dynamic
          re-confiduration of the cluster and track individual nodes
          going up and down. The code is pretty usable, we have it
          deployed for several customers and it's being actively
          developed as well. We have it working reliably controlling up
          to 30-40 RTPP instances scattered over at least 5 nodes.<br>
          <br>
          <a moz-do-not-send="true"
            href="http://sourceforge.net/p/sippy/sippy/ci/master/tree/rtp_cluster/">http://sourceforge.net/p/sippy/sippy/ci/master/tree/rtp_cluster/</a><br>
          <br>
        </div>
        We have at least one pretty well known service provider whose
        name starts with capital V using it in combination with OpenSIPS
        to load balance RTP traffic via bunch of Amazon EC2 instances.<br>
      </div>
      <div class="gmail_extra"><br>
        <br>
        <div class="gmail_quote">On Tue, May 27, 2014 at 6:52 AM, Brett
          Nemeroff <span dir="ltr">&lt;<a moz-do-not-send="true"
              href="mailto:brett@nemeroff.com" target="_blank">brett@nemeroff.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 dir="ltr">Just wanted to add my 0.02 here.. 
              <div><br>
              </div>
              <div>I totally agree with Bogdan. For the applications
                where opensips + a RTP relay make sense, HA and
                persistence are much more important. </div>
              <div><br>
              </div>
              <div>WebRTC and ICE are kinda applications in of
                themselves. And although these applications are going to
                grow in popularity, the "legacy" needs for an RTP relay
                are still massively prevalent in the space. A general
                push towards "Carrier Grade", resiliency and redundancy
                I think is much better for the project as a whole. </div>
              <div><br>
              </div>
              <div>Not only that, consider that applications requiring
                ICE or WebRTC will greatly benefit from HA /
                persistence, but not so much the other way around :) </div>
              <div><br>
              </div>
              <div>YMMV</div>
              <span class="HOEnZb"><font color="#888888">
                  <div>
                    <br>
                  </div>
                  <div>-Brett</div>
                  <div><br>
                  </div>
                </font></span></div>
            <div class="HOEnZb">
              <div class="h5">
                <div class="gmail_extra"><br>
                  <br>
                  <div class="gmail_quote">On Sun, May 25, 2014 at 6:30
                    AM, Bogdan-Andrei Iancu <span dir="ltr">&lt;<a
                        moz-do-not-send="true"
                        href="mailto:bogdan@opensips.org"
                        target="_blank">bogdan@opensips.org</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">
                        <div><tt>Hello,<br>
                            <br>
                            As always, the truth is in the middle.<br>
                            <br>
                            I agree RTPP is behind on certain things
                            (and this is why we want to do them), but on
                            the other hand it is a good platform with
                            other good features (missing on the other
                            relays). RTPP has better ability in
                            individually controlling the stream (audio
                            /video), ability to set timeouts and onhold
                            with no conflicts, ability to generates
                            events on timeout, more flexibility in
                            handling symmetric / asymmetric NATs,
                            ability to do media injection (playback),
                            ability to do call recording<br>
                            <br>
                            What neither  mediaproxy, nor rtpengine have
                            is a mechanism for implementing RTP failover
                            (for ongoing calls) or restart persistence .
                            This is something we want to look into. I
                            would love to have ICE and WebRTC on my
                            media relay, for the HA and persistence are
                            more important I would say.<br>
                            <br>
                            Regards,<br>
                          </tt>
                          <div>
                            <pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a moz-do-not-send="true" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre>
                          </div>
                          <div>
                            <div> On <a moz-do-not-send="true"
                                href="tel:24.05.2014%2001"
                                value="+12405201401" target="_blank">24.05.2014
                                01</a>:59, Muhammad Shahzad Shafi wrote:<br>
                            </div>
                          </div>
                        </div>
                        <blockquote type="cite">
                          <div>
                            <div>
                              <p>To be honest, i have stopped using
                                rtpproxy for over 2 years now. It is not
                                evolving as fast as it should be,
                                specially in the context of ICE and
                                WebRTC technologies.</p>
                              <p>I would like to suggest that opensips
                                team should consider adding support for
                                rtpengine from SIPWise,</p>
                              <p><a moz-do-not-send="true"
                                  href="https://github.com/sipwise/rtpengine"
                                  target="_blank">https://github.com/sipwise/rtpengine</a></p>
                              <p>For now mediaproxy from AG Projects is
                                the only good choice for handling media
                                in opensips with ICE support (though it
                                still lacks WebRTC features).</p>
                              <p>Thank you.</p>
                              <p> </p>
                              <p>On 2014-05-23 14:55, Bogdan-Andrei
                                Iancu wrote:</p>
                              <blockquote type="cite"
                                style="padding-left:5px;border-left:#1010ff
                                2px solid;margin-left:5px;width:100%"><tt>Going

                                  for a public exposure on this question
                                  to Maxim, maybe we will get an answer
                                  here.<br>
                                  <br>
                                </tt>
                                <div><br>
                                  -------- Original Message --------
                                  <table border="0" cellpadding="0"
                                    cellspacing="0">
                                    <tbody>
                                      <tr>
                                        <th align="RIGHT"
                                          nowrap="nowrap"
                                          valign="BASELINE">Subject:</th>
                                        <td>RTPproxy project</td>
                                      </tr>
                                      <tr>
                                        <th align="RIGHT"
                                          nowrap="nowrap"
                                          valign="BASELINE">Date:</th>
                                        <td>Mon, 14 Apr 2014 15:03:31
                                          +0300</td>
                                      </tr>
                                      <tr>
                                        <th align="RIGHT"
                                          nowrap="nowrap"
                                          valign="BASELINE">From:</th>
                                        <td>Bogdan-Andrei Iancu</td>
                                      </tr>
                                      <tr>
                                        <th align="RIGHT"
                                          nowrap="nowrap"
                                          valign="BASELINE">To:</th>
                                        <td>Maxim Sobolev</td>
                                      </tr>
                                      <tr>
                                        <th align="RIGHT"
                                          nowrap="nowrap"
                                          valign="BASELINE">CC:</th>
                                        <td>Razvan Crainea</td>
                                      </tr>
                                    </tbody>
                                  </table>
                                  <br>
                                  <br>
                                  <pre>Hello Maxim,

Long time, no talks, but I hope everything is fine on your side.

I'm reaching you in order to ask about your future plans in regards to 
the rtpproxy project? We see no much activity around it and other media 
relays are popping around.

RTPP is an essential component for us, we invested a lot of work, we 
have many patches (extensions) for it (which we want to push to the 
public tree, but there is no answer on this) and we are also looking for 
investing a lot into big future plans (as adding more functionalities).

Now, my question is - what is your commitment and disponibility for the 
RTPP project ? depending on that we what to re-position ourselves, as we 
do not want to waste time and work on things which are out of control.

Best regards,

-- 
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a moz-do-not-send="true" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a>


</pre>
                                </div>
                              </blockquote>
                              <div>
                                <pre>-- 
Mit freundlichen Grüßen
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: <a moz-do-not-send="true" href="tel:%2B49%20176%2099%2083%2010%2085" value="+4917699831085" target="_blank">+49 176 99 83 10 85</a>
MSN: <a moz-do-not-send="true" href="mailto:shari_786pk@hotmail.com" target="_blank">shari_786pk@hotmail.com</a>
Email: <a moz-do-not-send="true" href="mailto:shaheryarkh@googlemail.com" target="_blank">shaheryarkh@googlemail.com</a></pre>
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</pre>
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        <br>
        <br clear="all">
        <br>
        -- <br>
        <div dir="ltr">Maksym Sobolyev<br>
          Sippy Software, Inc.<br>
          Internet Telephony (VoIP) Experts<br>
          Tel (Canada): +1-778-783-0474<br>
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            href="mailto:sales@sippysoft.com" target="_blank">sales@sippysoft.com</a><br>
          Skype: SippySoft<br>
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