<div dir="ltr"><br><div class="gmail_extra"><br><div class="gmail_quote">On Mon, Sep 8, 2014 at 9:48 AM, Olli Heiskanen <span dir="ltr"><<a href="mailto:ohjelmistoarkkitehti@gmail.com" target="_blank">ohjelmistoarkkitehti@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>Hello,</div><div><br></div><div>I have a problem with a call between 2 webrtc clients. Asterisk removes the ice-related lines from the sdp when it sends the INVITE out, and the called webrtc client rejects the INVITE due to the missing ice lines. Both webrtc clients are defined exactly the same way, same values in all fields except the number of the peer.</div><div><br></div><div>There's probably something I've changed that causes this behavior. Can anyone tell me what's wrong in my configuration?</div><div><br></div><div><span style="font-family:arial,sans-serif;font-size:12.7272720336914px">res_rtp_asterisk is included in the compilation and uuid-devel is installed, Asterisk version is 11.11.0. Ice is enabled in rtp.conf as well as in both clients in the realtime sip peer table.</span><br></div><div><br></div><div>Here's my realtime peer data: </div><div>*CLI> realtime load sippeers name 660</div><div> Column Name Column Value</div><div> -------------------- --------------------</div><div> id 4</div><div> type friend</div><div> name 660</div><div> host dynamic</div><div> secret</div><div> encryption yes</div><div> avpf yes</div><div> icesupport yes <---- ICE is enabled</div><div> ipaddr PU.BL.IC.IP</div><div> port 5060</div><div> regseconds 1410185500</div><div> defaultuser 660</div><div> fullcontact sip:660@PU.BL.IC.IP:5060</div><div> lastms 0</div><div> useragent</div><div> context default</div><div> directmedia no</div><div> deny <a href="http://0.0.0.0/0.0.0.0" target="_blank">0.0.0.0/0.0.0.0</a></div><div> permit PU.BL.IC.IP</div><div> nat force_rport,comedia</div><div> language</div><div> disallow</div><div> allow</div><div> force_avp yes</div><div> callerid</div><div> amaflags</div><div> mailbox</div><div> regexten</div><div> regserver</div><div> fromdomain <a href="http://testers.com" target="_blank">testers.com</a></div><div> videosupport no</div><div> contactpermit</div><div> contactdeny</div><div> fullname 660 win8</div><div> hasvoicemail</div><div> subscribemwi</div><div> dtlsenable yes</div><div> dtlsverify no</div><div> dtlscertfile /etc/asterisk/keys/asterisk.pem</div><div> dtlsprivatekey /etc/asterisk/keys/asterisk.pem</div><div> dtlssetup actpass</div><div> sippasswd md5pwd</div><div> rpid</div><div> domain <a href="http://testers.com" target="_blank">testers.com</a></div><div> sippasswd2</div><div><br></div><div>and my sip.conf:</div><div><br></div><div>[general]</div><div>bindport = 5070</div><div>bindaddr = PU.BL.IC.IP</div><div>udpbindaddr = PU.BL.IC.IP</div><div>tcpenable = yes</div><div>limitonpeers = yes</div><div>rtcachefriends = no </div><div>tos_sip=cs3</div><div>tos_audio=ef</div><div>realm = <a href="http://testers.com" target="_blank">testers.com</a></div><div>autodomain=yes</div><div>domain=PU.BL.IC.IP</div><div>domain=<a href="http://testers.com" target="_blank">testers.com</a></div><div>transport=ws,wss,udp</div><div>outboundproxy=PU.BL.IC.IP:5060</div><div><br></div><div><br></div><div>I'd appreciate Your advice.</div><div><br></div><br></div></blockquote></div><br></div><div class="gmail_extra">What does a DEBUG log show with 'sip set debug on' when the outbound call is made?<br></div><div class="gmail_extra"><br>-- <br><div dir="ltr"><div>Matthew Jordan<br></div><div>Digium, Inc. | Engineering Manager</div><div>445 Jan Davis Drive NW - Huntsville, AL 35806 - USA</div><div>Check us out at: <a href="http://digium.com" target="_blank">http://digium.com</a> & <a href="http://asterisk.org" target="_blank">http://asterisk.org</a></div></div>
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