<div dir="ltr">Hi Razvan,<div><br></div><div>How to make follow connection using rtpengine?</div><div><br></div><div>Zoiper(g729) <-----> Opensips(rtpengine) <--------> browser (SIP.JS with g711) </div></div><div class="gmail_extra"><br><div class="gmail_quote">2017-04-18 19:10 GMT+03:00 Răzvan Crainea <span dir="ltr"><<a href="mailto:razvan@opensips.org" target="_blank">razvan@opensips.org</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
  
    
  
  <div bgcolor="#FFFFFF" text="#000000">
    <tt>Hi, Jeff!<br>
      <br>
      Unfortunately you can't use both rtpengine and codec_delete_*,
      that's because each change different buffers. The codec_delete_*
      function runs on the initial SDP received, then rtpengine
      completely overwrites the SDP with whatever rtpengine replied.<br>
      The only way you can do something like this (although it may be
      very ugly) is to store the rtpengine reply in a pvar using the
      3rd[1] parameter of the rtpengine_* functions and perform some
      text replaces[2] on it, then replace the body "manually".<br>
      <br>
      [1]
<a class="m_-8428955468467181929moz-txt-link-freetext" href="http://www.opensips.org/html/docs/modules/2.3.x/rtpengine.html#rtpengine.f.rtpengine_offer" target="_blank">http://www.opensips.org/html/<wbr>docs/modules/2.3.x/rtpengine.<wbr>html#rtpengine.f.rtpengine_<wbr>offer</a><br>
      [2]
      <a class="m_-8428955468467181929moz-txt-link-freetext" href="http://www.opensips.org/html/docs/modules/2.3.x/textops#idp5907728" target="_blank">http://www.opensips.org/html/<wbr>docs/modules/2.3.x/textops#<wbr>idp5907728</a><br>
      <br>
      Best regards,<br>
    </tt>
    <pre class="m_-8428955468467181929moz-signature" cols="72">Răzvan Crainea
OpenSIPS Solutions
<a class="m_-8428955468467181929moz-txt-link-abbreviated" href="http://www.opensips-solutions.com" target="_blank">www.opensips-solutions.com</a></pre>
    <div class="m_-8428955468467181929moz-cite-prefix">On 04/18/2017 06:49 PM, Jeff Pyle
      wrote:<br>
    </div>
    <blockquote type="cite">
      <div dir="ltr">Hello,
        <div><br>
        </div>
        <div>This is on OpenSIPS 2.3, downloaded from git and compiled
          today.</div>
        <div><br>
        </div>
        <div>An INVITE arrives over TLS with the following SDP:</div>
        <div><br>
        </div>
        <div>
          <div>v=0</div>
          <div>o=- 1492528621 1492528621 IN IP4 172.22.202.191</div>
          <div>s=Polycom IP Phone</div>
          <div>c=IN IP4 172.22.202.191</div>
          <div>t=0 0</div>
          <div>m=audio 16852 RTP/SAVP 115 9 0 8 110 18 127</div>
          <div>a=rtpmap:115 G7221/32000</div>
          <div>a=fmtp:115 bitrate=48000</div>
          <div>a=rtpmap:9 G722/8000</div>
          <div>a=rtpmap:0 PCMU/8000</div>
          <div>a=rtpmap:8 PCMA/8000</div>
          <div>a=rtpmap:110 iLBC/8000</div>
          <div>a=fmtp:110 mode=20</div>
          <div>a=rtpmap:18 G729/8000</div>
          <div>a=fmtp:18 annexb=no</div>
          <div>a=rtpmap:127 telephone-event/8000</div>
          <div>a=rtcp:16853</div>
          <div>a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:[stripped]</div>
          <div>a=setup:actpass</div>
          <div>a=fingerprint:sha-1 [stripped]</div>
          <div>m=audio 16888 RTP/AVP 115 9 0 8 110 18 127</div>
          <div>a=rtpmap:115 G7221/32000</div>
          <div>a=fmtp:115 bitrate=48000</div>
          <div>a=rtpmap:9 G722/8000</div>
          <div>a=rtpmap:0 PCMU/8000</div>
          <div>a=rtpmap:8 PCMA/8000</div>
          <div>a=rtpmap:110 iLBC/8000</div>
          <div>a=fmtp:110 mode=20</div>
          <div>a=rtpmap:18 G729/8000</div>
          <div>a=fmtp:18 annexb=no</div>
          <div>a=rtpmap:127 telephone-event/8000</div>
          <div>a=rtcp:16889</div>
        </div>
        <div><br>
        </div>
        <div>I run </div>
        <div>  codec_delete_expect_re(PCMU|<wbr>PCMA|telephone-event)</div>
        <div>but it doesn't have any effect.  The INVITE leaving after
          t_relay() over UDP to localhost on a different port is the
          same as when it came in (with the exception of the c= line
          because of rtpengine).</div>
        <div><br>
        </div>
        <div>At log_level=6 the only log entry I see is</div>
        <div>  DBG:sipmsgops:create_codec_<wbr>lumps: creating 0 streams</div>
        <div><br>
        </div>
        <div>I'm not sure where to go from here.</div>
        <div><br>
        </div>
        <div><br>
        </div>
        <div>- Jeff</div>
        <div><br>
        </div>
      </div>
      <br>
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      <br>
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    </blockquote>
    <br>
  </div>

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