<div dir="ltr">Dragomir, <div><br></div><div>If Zoiper speaks only G.729, and SIP.js speaks only G.711, rtpengine isn't going to help. It doesn't transcode. From its <a href="https://github.com/sipwise/rtpengine">github page</a>:</div><div><br></div><blockquote style="margin:0px 0px 0px 40px;border:none;padding:0px"><div><div><i>Rtpengine</i> does not (yet) support:</div></div></blockquote><div><div><ul><ul><li>Repacketization or transcoding</li></ul></ul></div></div><div><br></div><div>Is iLBC an option for you in SIP.js and Zoiper? It's license free and sounds a little bitter. If not, Asterisk or FreeSWITCH could perform this task with the appropriate G.729 licenses. </div><div><br></div><div><br></div><div><br></div><div><br></div><div>Răzvan,</div><div><br></div><div>Is there any effect of using either the codec manipulation or rtpengine in a branch route? I ask this admittedly not understanding the buffers in use.</div><div><br></div><div><br></div><div><br></div><div><br></div><div>- Jeff</div><div> </div><div><br></div><div><br></div><div><br></div><div><br><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Apr 18, 2017 at 12:39 PM, Dragomir Haralambiev <span dir="ltr"><<a href="mailto:goup2010@gmail.com" target="_blank">goup2010@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><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-HOEnZb"><div class="gmail-h5"><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:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div bgcolor="#FFFFFF">
<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="gmail-m_-1514856936150048012m_-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/d<wbr>ocs/modules/2.3.x/rtpengine.ht<wbr>ml#rtpengine.f.rtpengine_offer</a><br>
[2]
<a class="gmail-m_-1514856936150048012m_-8428955468467181929moz-txt-link-freetext" href="http://www.opensips.org/html/docs/modules/2.3.x/textops#idp5907728" target="_blank">http://www.opensips.org/html/d<wbr>ocs/modules/2.3.x/textops#idp5<wbr>907728</a><br>
<br>
Best regards,<br>
</tt>
<pre class="gmail-m_-1514856936150048012m_-8428955468467181929moz-signature" cols="72">Răzvan Crainea
OpenSIPS Solutions
<a class="gmail-m_-1514856936150048012m_-8428955468467181929moz-txt-link-abbreviated" href="http://www.opensips-solutions.com" target="_blank">www.opensips-solutions.com</a></pre>
<div class="gmail-m_-1514856936150048012m_-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|PC<wbr>MA|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_lum<wbr>ps: 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>
<fieldset class="gmail-m_-1514856936150048012m_-8428955468467181929mimeAttachmentHeader"></fieldset>
<br>
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</pre>
</blockquote>
<br>
</div>
</blockquote></div></div></div></div></blockquote></div><br></div></div></div>