<div dir="ltr">Hi Matthew,<div><br></div><div>Here's the debug output: </div><div><br></div><div><br></div><div><br></div><div><div><br></div><div><br></div><div><--- SIP read from UDP:PU.BL.IC.IP:5060 ---></div><div>INVITE <a href="mailto:sip%3A661@testers.com">sip:661@testers.com</a> SIP/2.0</div><div>Record-Route: <sip:PU.BL.IC.IP;r2=on;lr=on;ftag=856i7ei98p;nat=yes></div><div>Record-Route: <sip:PU.BL.IC.IP;transport=ws;r2=on;lr=on;ftag=856i7ei98p;nat=yes></div><div>Via: SIP/2.0/UDP PU.BL.IC.IP;branch=z9hG4bK3cb9.e83bde7c484eec5374f679e7232435af.0</div><div>Via: SIP/2.0/WS 8d833dsg4asm.invalid;rport=47184;received=CL.IE.NT.IP;branch=z9hG4bK3627044</div><div>Max-Forwards: 69</div><div>To: <<a href="mailto:sip%3A661@testers.com">sip:661@testers.com</a>></div><div>From: "660" <<a href="mailto:sip%3A660@testers.com">sip:660@testers.com</a>>;tag=856i7ei98p</div><div>Call-ID: oc0ppijresm05k2emsgt</div><div>CSeq: 3394 INVITE</div><div>Contact: <<a href="mailto:sip%3A660@testers.com">sip:660@testers.com</a>;gr=urn:uuid:81780308-9304-4e37-984f-a2e864b17bd3;alias=CL.IE.NT.IP~47184~5;alias=CL.IE.NT.IP~47184~5></div><div>Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE</div><div>Content-Type: application/sdp</div><div>Supported: gruu,outbound</div><div>User-Agent: SIP.js/0.6.2</div><div>Content-Length: 1862</div><div><br></div><div>v=0</div><div>o=- 9082254022026432015 2 IN IP4 PU.BL.IC.IP</div><div>s=-</div><div>t=0 0</div><div>a=group:BUNDLE audio</div><div>a=msid-semantic: WMS JtsXs3B8wXmgyUYVkUGWHBYVYgX0aSC38IWx</div><div>m=audio 10862 RTP/SAVPF 111 103 104 0 8 106 105 13 126</div><div>c=IN IP4 PU.BL.IC.IP</div><div>a=candidate:3350409123 1 udp 2122194687 192.168.0.101 65339 typ host generation 0</div><div>a=candidate:3350409123 2 udp 2122194687 192.168.0.101 65339 typ host generation 0</div><div>a=candidate:2301678419 1 tcp 1518214911 192.168.0.101 0 typ host generation 0</div><div>a=candidate:2301678419 2 tcp 1518214911 192.168.0.101 0 typ host generation 0</div><div>a=candidate:1190865175 1 udp 1685987071 CL.IE.NT.IP 65339 typ srflx raddr 192.168.0.101 rport 65339 generation 0</div><div>a=candidate:1190865175 2 udp 1685987071 CL.IE.NT.IP 65339 typ srflx raddr 192.168.0.101 rport 65339 generation 0</div><div>a=ice-ufrag:7N23UxBo9XUgx9pJ</div><div>a=ice-pwd:jL7AIeiJD5byGDSapfSftPRl</div><div>a=ice-options:google-ice</div><div>a=fingerprint:sha-256 93:E7:FC:E6:C2:74:71:2F:4F:81:43:7D:0C:A1:0F:C9:FC:3B:85:E6:44:2F:5A:39:05:79:DD:A6:0B:05:49:80</div><div>a=setup:actpass</div><div>a=mid:audio</div><div>a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level</div><div>a=extmap:3 <a href="http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time">http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time</a></div><div>a=rtpmap:111 opus/48000/2</div><div>a=fmtp:111 minptime=10</div><div>a=rtpmap:103 ISAC/16000</div><div>a=rtpmap:104 ISAC/32000</div><div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:8 PCMA/8000</div><div>a=rtpmap:106 CN/32000</div><div>a=rtpmap:105 CN/16000</div><div>a=rtpmap:13 CN/8000</div><div>a=rtpmap:126 telephone-event/8000</div><div>a=maxptime:60</div><div>a=ssrc:1842567493 cname:aJCyVX5iKPNU6Gf8</div><div>a=ssrc:1842567493 msid:JtsXs3B8wXmgyUYVkUGWHBYVYgX0aSC38IWx 01a46fec-8a85-412d-9905-dcbefb8952b6</div><div>a=ssrc:1842567493 mslabel:JtsXs3B8wXmgyUYVkUGWHBYVYgX0aSC38IWx</div><div>a=ssrc:1842567493 label:01a46fec-8a85-412d-9905-dcbefb8952b6</div><div>a=sendrecv</div><div>a=rtcp:10863</div><div>a=rtcp-mux</div><div>a=candidate:GjpvUWJxlvHL7PZ6 1 UDP 1518214655 PU.BL.IC.IP 10862 typ host</div><div>a=candidate:GjpvUWJxlvHL7PZ6 2 UDP 1518214654 PU.BL.IC.IP 10863 typ host</div><div><-------------></div><div>--- (16 headers 42 lines) ---</div><div>Sending to PU.BL.IC.IP:5060 (no NAT)</div><div>Sending to PU.BL.IC.IP:5060 (no NAT)</div><div>Using INVITE request as basis request - oc0ppijresm05k2emsgt</div><div>Found peer '660' for '660' from PU.BL.IC.IP:5060</div><div> == Using SIP RTP TOS bits 184</div><div> == Using SIP RTP CoS mark 5</div><div>Found RTP audio format 111</div><div>Found RTP audio format 103</div><div>Found RTP audio format 104</div><div>Found RTP audio format 0</div><div>Found RTP audio format 8</div><div>Found RTP audio format 106</div><div>Found RTP audio format 105</div><div>Found RTP audio format 13</div><div>Found RTP audio format 126</div><div>Found unknown media description format opus for ID 111</div><div>Found unknown media description format ISAC for ID 103</div><div>Found unknown media description format ISAC for ID 104</div><div>Found audio description format PCMU for ID 0</div><div>Found audio description format PCMA for ID 8</div><div>Found unknown media description format CN for ID 106</div><div>Found unknown media description format CN for ID 105</div><div>Found audio description format CN for ID 13</div><div>Found audio description format telephone-event for ID 126</div><div>Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)</div><div>Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)</div><div>Peer audio RTP is at port PU.BL.IC.IP:10862</div><div>Looking for 661 in default (domain <a href="http://testers.com">testers.com</a>)</div><div>list_route: hop: <sip:PU.BL.IC.IP;r2=on;lr=on;ftag=856i7ei98p;nat=yes></div><div>list_route: hop: <sip:PU.BL.IC.IP;transport=ws;r2=on;lr=on;ftag=856i7ei98p;nat=yes></div><div><br></div><div><--- Transmitting (NAT) to PU.BL.IC.IP:5060 ---></div><div>SIP/2.0 100 Trying</div><div>Via: SIP/2.0/UDP PU.BL.IC.IP;branch=z9hG4bK3cb9.e83bde7c484eec5374f679e7232435af.0;received=PU.BL.IC.IP;rport=5060</div><div>Via: SIP/2.0/WS 8d833dsg4asm.invalid;rport=47184;received=CL.IE.NT.IP;branch=z9hG4bK3627044</div><div>Record-Route: <sip:PU.BL.IC.IP;r2=on;lr=on;ftag=856i7ei98p;nat=yes></div><div>Record-Route: <sip:PU.BL.IC.IP;transport=ws;r2=on;lr=on;ftag=856i7ei98p;nat=yes></div><div>From: "660" <<a href="mailto:sip%3A660@testers.com">sip:660@testers.com</a>>;tag=856i7ei98p</div><div>To: <<a href="mailto:sip%3A661@testers.com">sip:661@testers.com</a>></div><div>Call-ID: oc0ppijresm05k2emsgt</div><div>CSeq: 3394 INVITE</div><div>Server: I Am the Devil</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE</div><div>Supported: replaces, timer</div><div>Contact: <sip:661@PU.BL.IC.IP:5070></div><div>Content-Length: 0</div><div><br></div><div><br></div><div><------------></div><div> -- Executing [661@default:1] NoOp("SIP/660-00000007", "general : Dialed 661") in new stack</div><div> -- Executing [661@default:2] Dial("SIP/660-00000007", "SIP/661,3600,rt") in new stack</div><div> == Using SIP RTP TOS bits 184</div><div> == Using SIP RTP CoS mark 5</div><div>Audio is at 18366</div><div>Adding codec 100003 (ulaw) to SDP</div><div>Adding codec 100002 (gsm) to SDP</div><div>Adding codec 100004 (alaw) to SDP</div><div>Adding codec 100017 (testlaw) to SDP</div><div>Adding non-codec 0x1 (telephone-event) to SDP</div><div>Reliably Transmitting (NAT) to PU.BL.IC.IP:5060:</div><div>INVITE sip:661@PU.BL.IC.IP:5060 SIP/2.0</div><div>Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK4e738afa;rport</div><div>Max-Forwards: 70</div><div>From: "660 win8" <<a href="mailto:sip%3A660@testers.com">sip:660@testers.com</a>>;tag=as73376885</div><div>To: <sip:661@PU.BL.IC.IP:5060></div><div>Contact: <sip:660@PU.BL.IC.IP:5070></div><div>Call-ID: <a href="mailto:2f70cc9567be50a46ba2879d4391a7dc@testers.com">2f70cc9567be50a46ba2879d4391a7dc@testers.com</a></div><div>CSeq: 102 INVITE</div><div>User-Agent: I Am the Devil</div><div>Date: Mon, 08 Sep 2014 15:15:37 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: 437</div><div><br></div><div>v=0</div><div>o=root 630896079 630896079 IN IP4 PU.BL.IC.IP</div><div>s=Asterisk PBX 11.11.0</div><div>c=IN IP4 PU.BL.IC.IP</div><div>t=0 0</div><div>m=audio 18366 RTP/SAVPF 0 3 8 101</div><div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:3 GSM/8000</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=connection:new</div><div>a=setup:actpass</div><div>a=fingerprint:SHA-256 CE:EE:D9:28:EA:B0:6E:D0:CE:4F:5A:9A:FB:53:66:74:83:47:18:37:2F:76:C1:6D:10:C0:EE:FF:A4:56:F4:05</div><div>a=sendrecv</div><div><br></div><div>---</div><div><br></div><div><--- SIP read from UDP:PU.BL.IC.IP:5060 ---></div><div>SIP/2.0 100 trying -- your call is important to us</div><div>Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK4e738afa;rport=5070</div><div>From: "660 win8" <<a href="mailto:sip%3A660@testers.com">sip:660@testers.com</a>>;tag=as73376885</div><div>To: <sip:661@PU.BL.IC.IP:5060></div><div>Call-ID: <a href="mailto:2f70cc9567be50a46ba2879d4391a7dc@testers.com">2f70cc9567be50a46ba2879d4391a7dc@testers.com</a></div><div>CSeq: 102 INVITE</div><div>Content-Length: 0</div><div><br></div><div><-------------></div><div>--- (7 headers 0 lines) ---</div><div> -- Called SIP/661</div><div><br></div><div><--- Transmitting (NAT) to PU.BL.IC.IP:5060 ---></div><div>SIP/2.0 180 Ringing</div><div>Via: SIP/2.0/UDP PU.BL.IC.IP;branch=z9hG4bK3cb9.e83bde7c484eec5374f679e7232435af.0;received=PU.BL.IC.IP;rport=5060</div><div>Via: SIP/2.0/WS 8d833dsg4asm.invalid;rport=47184;received=CL.IE.NT.IP;branch=z9hG4bK3627044</div><div>Record-Route: <sip:PU.BL.IC.IP;r2=on;lr=on;ftag=856i7ei98p;nat=yes></div><div>Record-Route: <sip:PU.BL.IC.IP;transport=ws;r2=on;lr=on;ftag=856i7ei98p;nat=yes></div><div>From: "660" <<a href="mailto:sip%3A660@testers.com">sip:660@testers.com</a>>;tag=856i7ei98p</div><div>To: <<a href="mailto:sip%3A661@testers.com">sip:661@testers.com</a>>;tag=as4298ec2e</div><div>Call-ID: oc0ppijresm05k2emsgt</div><div>CSeq: 3394 INVITE</div><div>Server: I Am the Devil</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE</div><div>Supported: replaces, timer</div><div>Contact: <sip:661@PU.BL.IC.IP:5070></div><div>Content-Length: 0</div><div><br></div><div><br></div><div><------------></div><div><br></div><div><--- SIP read from UDP:PU.BL.IC.IP:5060 ---></div><div>SIP/2.0 404 No destinations</div><div>Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK4e738afa;rport=5070</div><div>From: "660 win8" <<a href="mailto:sip%3A660@testers.com">sip:660@testers.com</a>>;tag=as73376885</div><div>To: <sip:661@PU.BL.IC.IP:5060>;tag=b552f34cdfad88fd2d6dc20c55c3a3ed-ba78</div><div>Call-ID: <a href="mailto:2f70cc9567be50a46ba2879d4391a7dc@testers.com">2f70cc9567be50a46ba2879d4391a7dc@testers.com</a></div><div>CSeq: 102 INVITE</div><div>Content-Length: 0</div><div><br></div><div><-------------></div><div>--- (7 headers 0 lines) ---</div><div>Transmitting (NAT) to PU.BL.IC.IP:5060:</div><div>ACK sip:661@PU.BL.IC.IP:5060 SIP/2.0</div><div>Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK4e738afa;rport</div><div>Max-Forwards: 70</div><div>From: "660 win8" <<a href="mailto:sip%3A660@testers.com">sip:660@testers.com</a>>;tag=as73376885</div><div>To: <sip:661@PU.BL.IC.IP:5060>;tag=b552f34cdfad88fd2d6dc20c55c3a3ed-ba78</div><div>Contact: <sip:660@PU.BL.IC.IP:5070></div><div>Call-ID: <a href="mailto:2f70cc9567be50a46ba2879d4391a7dc@testers.com">2f70cc9567be50a46ba2879d4391a7dc@testers.com</a></div><div>CSeq: 102 ACK</div><div>User-Agent: I Am the Devil</div><div>Content-Length: 0</div><div><br></div><div><br></div><div>---</div><div>Scheduling destruction of SIP dialog '<a href="mailto:2f70cc9567be50a46ba2879d4391a7dc@testers.com">2f70cc9567be50a46ba2879d4391a7dc@testers.com</a>' in 32000 ms (Method: INVITE)</div><div> == Everyone is busy/congested at this time (1:0/0/1)</div><div> -- Executing [661@default:3] Hangup("SIP/660-00000007", "") in new stack</div><div> == Spawn extension (default, 661, 3) exited non-zero on 'SIP/660-00000007'</div><div>Scheduling destruction of SIP dialog 'oc0ppijresm05k2emsgt' in 32000 ms (Method: INVITE)</div><div><br></div><div><--- Reliably Transmitting (NAT) to PU.BL.IC.IP:5060 ---></div><div>SIP/2.0 404 Not Found</div><div>Via: SIP/2.0/UDP PU.BL.IC.IP;branch=z9hG4bK3cb9.e83bde7c484eec5374f679e7232435af.0;received=PU.BL.IC.IP;rport=5060</div><div>Via: SIP/2.0/WS 8d833dsg4asm.invalid;rport=47184;received=CL.IE.NT.IP;branch=z9hG4bK3627044</div><div>From: "660" <<a href="mailto:sip%3A660@testers.com">sip:660@testers.com</a>>;tag=856i7ei98p</div><div>To: <<a href="mailto:sip%3A661@testers.com">sip:661@testers.com</a>>;tag=as4298ec2e</div><div>Call-ID: oc0ppijresm05k2emsgt</div><div>CSeq: 3394 INVITE</div><div>Server: I Am the Devil</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE</div><div>Supported: replaces, timer</div><div>Content-Length: 0</div><div><br></div><div><br></div><div><------------></div><div><br></div><div><--- SIP read from UDP:PU.BL.IC.IP:5060 ---></div><div>ACK <a href="mailto:sip%3A661@testers.com">sip:661@testers.com</a> SIP/2.0</div><div>Via: SIP/2.0/UDP PU.BL.IC.IP;branch=z9hG4bK3cb9.e83bde7c484eec5374f679e7232435af.0</div><div>Max-Forwards: 69</div><div>To: <<a href="mailto:sip%3A661@testers.com">sip:661@testers.com</a>>;tag=as4298ec2e</div><div>From: "660" <<a href="mailto:sip%3A660@testers.com">sip:660@testers.com</a>>;tag=856i7ei98p</div><div>Call-ID: oc0ppijresm05k2emsgt</div><div>CSeq: 3394 ACK</div><div>Content-Length: 0</div><div><br></div><div><-------------></div><div>--- (8 headers 0 lines) ---</div><div>u363id562*CLI></div></div><div><br></div><div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">2014-09-08 17:57 GMT+03:00 Matthew Jordan <span dir="ltr"><<a href="mailto:mjordan@digium.com" target="_blank">mjordan@digium.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"><div><div class="h5"><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></div><div class="gmail_extra">What does a DEBUG log show with 'sip set debug on' when the outbound call is made?<span class="HOEnZb"><font color="#888888"><br></font></span></div><span class="HOEnZb"><font color="#888888"><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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