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<tt>Hi, Jeff!<br>
<br>
Unfortunately I don't think this will work.<br>
The idea is that during a message processing, the same SDP buffer
is seen by all functions. Changes do not immediately alter the
buffer, but they are registered as lumps (changes that are later
applied), when the message processing is done. So even in the
branch route you will see the initial SDP, because it is part of
the same processing context (branch route is triggered by the
t_relay() function).<br>
<br>
Best regards,<br>
</tt>
<pre class="moz-signature" cols="72">Răzvan Crainea
OpenSIPS Solutions
<a class="moz-txt-link-abbreviated" href="http://www.opensips-solutions.com">www.opensips-solutions.com</a></pre>
<div class="moz-cite-prefix">On 04/18/2017 09:44 PM, Jeff Pyle
wrote:<br>
</div>
<blockquote
cite="mid:CAPhW+0+VMxkNpmgnTtGbP0iKrNUoOTNnkKrC7ULevSHE2C-sag@mail.gmail.com"
type="cite">
<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 moz-do-not-send="true"
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
moz-do-not-send="true"
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
moz-do-not-send="true"
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 moz-do-not-send="true"
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 moz-do-not-send="true"
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 moz-do-not-send="true" 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>
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</blockquote>
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