<div dir="ltr">And what returns prtpengine at log when changing this packet.<br><br><div>Returning to SIP proxy: d3:sdp316:v=0#015#012o=root 1195474335 1195474335 IN IP4 2.10.39.16#015#012s=Asterisk PBX 12.6.1#015#012c=IN IP4 2.10.39.16#015#012t=0 0#015#012m=audio 30614 RTP/AVP 8 3 0 101#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:3 GSM/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=ptime:20#015#012a=maxptime:150#015#012a=sendrecv#015#012a=rtcp:30615#015#0126:result2:oke</div><div><br></div><div>So it looks like that Destination sets from second append_branch at second step (to UDP)  and body sets as body of first step (for WS packet) </div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">2014-10-23 23:36 GMT+04:00 Yuriy Gorlichenko <span dir="ltr"><<a href="mailto:ovoshlook@gmail.com" target="_blank">ovoshlook@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">No SDP body only one. but packet like this<br><br><div>INVITE sip:device-200@sip:1.21.10.2:45437;rinstance=07f88c423145358e;transport=UDP SIP/2.0</div><div>Record-Route: <sip:sip.myservice.com:5068;nat=yes;ftag=as1be940e5;lr=on></div><div>Via: SIP/2.0/UDP sip.myservice.com:5068;branch=z9hG4bKca7d.2d16143316e23fac46bf686bb41780b3.2</div><div>Via: SIP/2.0/UDP 17.74.28.7:50600;branch=z9hG4bK22c67800;rport=50600</div><div>Max-Forwards: 70</div><div>From: "Name" <<a href="http://sip:1001@17.74.28.7:50600" target="_blank">sip:1001@17.74.28.7:50600</a>>;tag=as1be940e5</div><div>To: <<a href="http://sip:device-200@sip.myservice.com:5068" target="_blank">sip:device-200@sip.myservice.com:5068</a>></div><div>Contact: <<a href="http://sip:1001@17.74.28.7:50600" target="_blank">sip:1001@17.74.28.7:50600</a>></div><div>Call-ID: <a href="http://5ee58acd136888261e85d91e345e7ba1@17.74.28.7:50600" target="_blank">5ee58acd136888261e85d91e345e7ba1@17.74.28.7:50600</a></div><div>CSeq: 102 INVITE</div><div>User-Agent: Asterisk PBX 12.6.1</div><div>Date: Thu, 23 Oct 2014 19:27:54 GMT</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE</div><div>Supported: replaces, timer</div><div>Content-Type: application/sdp</div><div>Content-Length: 1044</div><div><br></div><div>v=0</div><div>o=root 1195474335 1195474335 IN IP4 2.10.39.16</div><div>s=Asterisk PBX 12.6.1</div><div>c=IN IP4 2.10.39.16</div><div>t=0 0</div><div>a=ice-lite</div><div>m=audio 30614 RTP/SAVPF 8 3 0 101</div><div>a=rtpmap:8 PCMA/8000</div><div>a=rtpmap:3 GSM/8000</div><div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div><div>a=ptime:20</div><div>a=maxptime:150</div><div>a=sendrecv</div><div>a=rtcp:30615</div><div>a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:OY72ZDHa+E3avlHwschrdBMe00qDfkN0BUyOxT1C</div><div>a=setup:actpass</div><div>a=fingerprint:sha-1 07:3D:B4:B0:0E:0D:87:39:C3:83:10:E2:B8:B8:2C:0C:0D:59:EF:4C</div><div>a=ice-ufrag:Wudfwh08</div><div>a=ice-pwd:VoamuFVRrAXOhUaeD6tA3PcXhndL</div><div>a=candidate:8jYonvAy1KGkAdP3 1 UDP 213070</div><div><br></div><br><br></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">2014-10-23 23:25 GMT+04:00 Richard Fuchs <span dir="ltr"><<a href="mailto:rfuchs@sipwise.com" target="_blank">rfuchs@sipwise.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span>On 10/23/14 15:06, Yuriy Gorlichenko wrote:<br>
> Still have same error...<br>
> Now rtpproxy_manage("co-sp") for classic call. At log I see that<br>
> rtpproxy wirked gine. For each step it generate write body, but t_Relay<br>
> still send strange "compinated" packet to UDP with SDP for WS...<br>
<br>
</span>Do you mean that the outgoing packet contains two SDP bodies? This has<br>
been discussed and solved in this thread:<br>
<a href="http://lists.sip-router.org/pipermail/sr-dev/2014-July/024507.html" target="_blank">http://lists.sip-router.org/pipermail/sr-dev/2014-July/024507.html</a><br>
<div><div><br>
cheers<br>
<br>
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</div></div></blockquote></div><br></div>
</div></div></blockquote></div><br></div>