<div dir="ltr">Hello,<div><br></div><div>i'm facing a strange issue on a trunk (on Internet). Here is the summary of the call flow :</div><div>1. The call is established normally (RTP Ports, Codec). Everything is fine.</div><div>2. During the call, Freeswitch receives a SIP OPTIONS with the SAME callid as the ongoing call.</div><div>3. Freeswitch answers 200 OK, but with a SDP with a 0 RTP port.</div><div>4. Our facing SIP Proxy uses 0 for next RTP trafic and then the communication continues with no sound</div><div><br></div><div>Is this a normal behaviour from Freeswitch ?</div><div>Do I have a possible configuration item to tune for that ?</div><div><br></div><div>Here is the SIP OPTIONS part of the SIP trace :</div><div><br></div><div> ------------------------------------------------------------------------</div><div>recv 436 bytes from udp/[PUBLIC_SOURCE_IP]:5060 at 17:20:58.284455:</div><div> ------------------------------------------------------------------------</div><div> OPTIONS sip:+3310129144152000@10.9.16.171:5060;transport=udp SIP/2.0</div><div> Via: SIP/2.0/UDP PUBLIC_SOURCE_IP:5060;branch=z9hG4bK72be.d4033057.0</div><div> To: <<a href="mailto:sip%3A%2B3310129144152000@domain.net">sip:+3310129144152000@domain.net</a>;user=phone>;tag=2tHZ5Xgv3HB8c</div><div> From: <<a href="mailto:sip%3A%2B33624431415@domain.net">sip:+33624431415@domain.net</a>;user=phone>;tag=SD66qua01-7e4e5aa0-0019-088d-0000-0000</div><div> CSeq: 2 OPTIONS</div><div> Call-ID: SD66qua01-64e828d3c9030b4024d0e46628fb2949-0000000</div><div> Max-Forwards: 70</div><div> Content-Length: 0</div><div> User-Agent: Keyyo-S5-P</div><div><br></div><div> ------------------------------------------------------------------------</div><div>send 905 bytes to udp/[PUBLIC_SOURCE_IP]:5060 at 17:20:58.289599:</div><div> ------------------------------------------------------------------------</div><div> SIP/2.0 200 OK</div><div> Via: SIP/2.0/UDP PUBLIC_SOURCE_IP:5060;branch=z9hG4bK72be.d4033057.0</div><div> From: <<a href="mailto:sip%3A%2B33624431415@domain.net">sip:+33624431415@domain.net</a>;user=phone>;tag=SD66qua01-7e4e5aa0-0019-088d-0000-0000</div><div> To: <<a href="mailto:sip%3A%2B3310129144152000@domain.net">sip:+3310129144152000@domain.net</a>;user=phone>;tag=2tHZ5Xgv3HB8c</div><div> Call-ID: SD66qua01-64e828d3c9030b4024d0e46628fb2949-0000000</div><div> CSeq: 2 OPTIONS</div><div> Contact: <sip:+3310129144152000@FREESWITCH_PUBLIC_IP:5060;transport=udp></div><div> User-Agent: FreeSWITCH</div><div> Accept: application/sdp</div><div> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY</div><div> Supported: timer, path, replaces</div><div> Allow-Events: talk, hold, conference, refer</div><div> Content-Type: application/sdp</div><div> Content-Length: 256</div><div><br></div><div> v=0</div><div> o=FreeSWITCH 1476957466 1476957467 IN IP4 FREESWITCH_PUBLIC_IP</div><div> s=FreeSWITCH</div><div> c=IN IP4 FREESWITCH_PUBLIC_IP</div><div> t=0 0</div><div> m=audio<b> <font color="#ff0000">0</font></b> RTP/AVP 8 101</div><div> a=rtpmap:8 PCMA/8000</div><div> a=rtpmap:101 telephone-event/8000</div><div> a=fmtp:101 0-16</div><div> a=ptime:20</div><div> a=rtcp:19369 IN IP4 FREESWITCH_PUBLIC_IP</div><div> ------------------------------------------------------------------------ </div><div><div class="gmail_extra"><br clear="all"><div><div class="gmail_signature"><div dir="ltr"><div><p><font face="verdana, sans-serif"><b><span lang="EN-US"><span></span><span></span>Dominique
Jeannerod</span></b><span lang="EN-US"></span></font></p>
<p><span lang="EN-US"><i><font face="verdana, sans-serif">Interact-iv.com</font></i></span></p><font face="verdana, sans-serif"><font size="1">Mobile: +33 698 778 763</font><br><font size="1"><br></font></font><div><font face="verdana, sans-serif"><font size="1">Headquarter : 600 rue Felix Trombe – tecnosud – 66100 Perpignan - France</font><br><font size="1">France : 0826 10 23 21 (0,15€/mn) International +33 (0)4 20 10 45 45</font></font></div></div></div></div></div></div></div></div>