<html>
<head>
<meta content="text/html; charset=utf-8" http-equiv="Content-Type">
</head>
<body 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="moz-txt-link-freetext" href="http://www.opensips.org/html/docs/modules/2.3.x/rtpengine.html#rtpengine.f.rtpengine_offer">http://www.opensips.org/html/docs/modules/2.3.x/rtpengine.html#rtpengine.f.rtpengine_offer</a><br>
[2]
<a class="moz-txt-link-freetext" href="http://www.opensips.org/html/docs/modules/2.3.x/textops#idp5907728">http://www.opensips.org/html/docs/modules/2.3.x/textops#idp5907728</a><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 06:49 PM, Jeff Pyle
wrote:<br>
</div>
<blockquote
cite="mid:CAPhW+0Lpv4LK8gOR_6w8jJfxnASaU74xrMN3ZXD6NJ_sJEhdHQ@mail.gmail.com"
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|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_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>
<fieldset class="mimeAttachmentHeader"></fieldset>
<br>
<pre wrap="">_______________________________________________
Users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a>
<a class="moz-txt-link-freetext" href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a>
</pre>
</blockquote>
<br>
</body>
</html>