Hi Matthew, Here's the debug output:
<--- SIP read from UDP:PU.BL.IC.IP:5060 ---> INVITE sip:[email protected] SIP/2.0 Record-Route: <sip:PU.BL.IC.IP;r2=on;lr=on;ftag=856i7ei98p;nat=yes> Record-Route: <sip:PU.BL.IC.IP;transport=ws;r2=on;lr=on;ftag=856i7ei98p;nat=yes> Via: SIP/2.0/UDP PU.BL.IC.IP;branch=z9hG4bK3cb9.e83bde7c484eec5374f679e7232435af.0 Via: SIP/2.0/WS 8d833dsg4asm.invalid;rport=47184;received=CL.IE.NT.IP;branch=z9hG4bK3627044 Max-Forwards: 69 To: <sip:[email protected]> From: "660" <sip:[email protected]>;tag=856i7ei98p Call-ID: oc0ppijresm05k2emsgt CSeq: 3394 INVITE Contact: <sip:[email protected] ;gr=urn:uuid:81780308-9304-4e37-984f-a2e864b17bd3;alias=CL.IE.NT.IP~47184~5;alias=CL.IE.NT.IP~47184~5> Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE Content-Type: application/sdp Supported: gruu,outbound User-Agent: SIP.js/0.6.2 Content-Length: 1862 v=0 o=- 9082254022026432015 2 IN IP4 PU.BL.IC.IP s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS JtsXs3B8wXmgyUYVkUGWHBYVYgX0aSC38IWx m=audio 10862 RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 PU.BL.IC.IP a=candidate:3350409123 1 udp 2122194687 192.168.0.101 65339 typ host generation 0 a=candidate:3350409123 2 udp 2122194687 192.168.0.101 65339 typ host generation 0 a=candidate:2301678419 1 tcp 1518214911 192.168.0.101 0 typ host generation 0 a=candidate:2301678419 2 tcp 1518214911 192.168.0.101 0 typ host generation 0 a=candidate:1190865175 1 udp 1685987071 CL.IE.NT.IP 65339 typ srflx raddr 192.168.0.101 rport 65339 generation 0 a=candidate:1190865175 2 udp 1685987071 CL.IE.NT.IP 65339 typ srflx raddr 192.168.0.101 rport 65339 generation 0 a=ice-ufrag:7N23UxBo9XUgx9pJ a=ice-pwd:jL7AIeiJD5byGDSapfSftPRl a=ice-options:google-ice 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 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:1842567493 cname:aJCyVX5iKPNU6Gf8 a=ssrc:1842567493 msid:JtsXs3B8wXmgyUYVkUGWHBYVYgX0aSC38IWx 01a46fec-8a85-412d-9905-dcbefb8952b6 a=ssrc:1842567493 mslabel:JtsXs3B8wXmgyUYVkUGWHBYVYgX0aSC38IWx a=ssrc:1842567493 label:01a46fec-8a85-412d-9905-dcbefb8952b6 a=sendrecv a=rtcp:10863 a=rtcp-mux a=candidate:GjpvUWJxlvHL7PZ6 1 UDP 1518214655 PU.BL.IC.IP 10862 typ host a=candidate:GjpvUWJxlvHL7PZ6 2 UDP 1518214654 PU.BL.IC.IP 10863 typ host <-------------> --- (16 headers 42 lines) --- Sending to PU.BL.IC.IP:5060 (no NAT) Sending to PU.BL.IC.IP:5060 (no NAT) Using INVITE request as basis request - oc0ppijresm05k2emsgt Found peer '660' for '660' from PU.BL.IC.IP:5060 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Found RTP audio format 111 Found RTP audio format 103 Found RTP audio format 104 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 106 Found RTP audio format 105 Found RTP audio format 13 Found RTP audio format 126 Found unknown media description format opus for ID 111 Found unknown media description format ISAC for ID 103 Found unknown media description format ISAC for ID 104 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found unknown media description format CN for ID 106 Found unknown media description format CN for ID 105 Found audio description format CN for ID 13 Found audio description format telephone-event for ID 126 Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|) Peer audio RTP is at port PU.BL.IC.IP:10862 Looking for 661 in default (domain testers.com) list_route: hop: <sip:PU.BL.IC.IP;r2=on;lr=on;ftag=856i7ei98p;nat=yes> list_route: hop: <sip:PU.BL.IC.IP;transport=ws;r2=on;lr=on;ftag=856i7ei98p;nat=yes> <--- Transmitting (NAT) to PU.BL.IC.IP:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP PU.BL.IC.IP;branch=z9hG4bK3cb9.e83bde7c484eec5374f679e7232435af.0;received=PU.BL.IC.IP;rport=5060 Via: SIP/2.0/WS 8d833dsg4asm.invalid;rport=47184;received=CL.IE.NT.IP;branch=z9hG4bK3627044 Record-Route: <sip:PU.BL.IC.IP;r2=on;lr=on;ftag=856i7ei98p;nat=yes> Record-Route: <sip:PU.BL.IC.IP;transport=ws;r2=on;lr=on;ftag=856i7ei98p;nat=yes> From: "660" <sip:[email protected]>;tag=856i7ei98p To: <sip:[email protected]> Call-ID: oc0ppijresm05k2emsgt CSeq: 3394 INVITE Server: I Am the Devil Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:[email protected]:5070> Content-Length: 0 <------------> -- Executing [661@default:1] NoOp("SIP/660-00000007", "general : Dialed 661") in new stack -- Executing [661@default:2] Dial("SIP/660-00000007", "SIP/661,3600,rt") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Audio is at 18366 Adding codec 100003 (ulaw) to SDP Adding codec 100002 (gsm) to SDP Adding codec 100004 (alaw) to SDP Adding codec 100017 (testlaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to PU.BL.IC.IP:5060: INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK4e738afa;rport Max-Forwards: 70 From: "660 win8" <sip:[email protected]>;tag=as73376885 To: <sip:[email protected]:5060> Contact: <sip:[email protected]:5070> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: I Am the Devil Date: Mon, 08 Sep 2014 15:15:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 437 v=0 o=root 630896079 630896079 IN IP4 PU.BL.IC.IP s=Asterisk PBX 11.11.0 c=IN IP4 PU.BL.IC.IP t=0 0 m=audio 18366 RTP/SAVPF 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=connection:new a=setup:actpass 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 a=sendrecv --- <--- SIP read from UDP:PU.BL.IC.IP:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK4e738afa;rport=5070 From: "660 win8" <sip:[email protected]>;tag=as73376885 To: <sip:[email protected]:5060> Call-ID: [email protected] CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- -- Called SIP/661 <--- Transmitting (NAT) to PU.BL.IC.IP:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP PU.BL.IC.IP;branch=z9hG4bK3cb9.e83bde7c484eec5374f679e7232435af.0;received=PU.BL.IC.IP;rport=5060 Via: SIP/2.0/WS 8d833dsg4asm.invalid;rport=47184;received=CL.IE.NT.IP;branch=z9hG4bK3627044 Record-Route: <sip:PU.BL.IC.IP;r2=on;lr=on;ftag=856i7ei98p;nat=yes> Record-Route: <sip:PU.BL.IC.IP;transport=ws;r2=on;lr=on;ftag=856i7ei98p;nat=yes> From: "660" <sip:[email protected]>;tag=856i7ei98p To: <sip:[email protected]>;tag=as4298ec2e Call-ID: oc0ppijresm05k2emsgt CSeq: 3394 INVITE Server: I Am the Devil Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:[email protected]:5070> Content-Length: 0 <------------> <--- SIP read from UDP:PU.BL.IC.IP:5060 ---> SIP/2.0 404 No destinations Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK4e738afa;rport=5070 From: "660 win8" <sip:[email protected]>;tag=as73376885 To: <sip:[email protected]:5060>;tag=b552f34cdfad88fd2d6dc20c55c3a3ed-ba78 Call-ID: [email protected] CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Transmitting (NAT) to PU.BL.IC.IP:5060: ACK sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK4e738afa;rport Max-Forwards: 70 From: "660 win8" <sip:[email protected]>;tag=as73376885 To: <sip:[email protected]:5060>;tag=b552f34cdfad88fd2d6dc20c55c3a3ed-ba78 Contact: <sip:[email protected]:5070> Call-ID: [email protected] CSeq: 102 ACK User-Agent: I Am the Devil Content-Length: 0 --- Scheduling destruction of SIP dialog ' [email protected]' in 32000 ms (Method: INVITE) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [661@default:3] Hangup("SIP/660-00000007", "") in new stack == Spawn extension (default, 661, 3) exited non-zero on 'SIP/660-00000007' Scheduling destruction of SIP dialog 'oc0ppijresm05k2emsgt' in 32000 ms (Method: INVITE) <--- Reliably Transmitting (NAT) to PU.BL.IC.IP:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP PU.BL.IC.IP;branch=z9hG4bK3cb9.e83bde7c484eec5374f679e7232435af.0;received=PU.BL.IC.IP;rport=5060 Via: SIP/2.0/WS 8d833dsg4asm.invalid;rport=47184;received=CL.IE.NT.IP;branch=z9hG4bK3627044 From: "660" <sip:[email protected]>;tag=856i7ei98p To: <sip:[email protected]>;tag=as4298ec2e Call-ID: oc0ppijresm05k2emsgt CSeq: 3394 INVITE Server: I Am the Devil Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> <--- SIP read from UDP:PU.BL.IC.IP:5060 ---> ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP PU.BL.IC.IP;branch=z9hG4bK3cb9.e83bde7c484eec5374f679e7232435af.0 Max-Forwards: 69 To: <sip:[email protected]>;tag=as4298ec2e From: "660" <sip:[email protected]>;tag=856i7ei98p Call-ID: oc0ppijresm05k2emsgt CSeq: 3394 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- u363id562*CLI> 2014-09-08 17:57 GMT+03:00 Matthew Jordan <[email protected]>: > > > On Mon, Sep 8, 2014 at 9:48 AM, Olli Heiskanen < > [email protected]> wrote: > >> Hello, >> >> 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. >> >> There's probably something I've changed that causes this behavior. Can >> anyone tell me what's wrong in my configuration? >> >> 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. >> >> Here's my realtime peer data: >> *CLI> realtime load sippeers name 660 >> Column Name Column Value >> -------------------- -------------------- >> id 4 >> type friend >> name 660 >> host dynamic >> secret >> encryption yes >> avpf yes >> icesupport yes <---- ICE is enabled >> ipaddr PU.BL.IC.IP >> port 5060 >> regseconds 1410185500 >> defaultuser 660 >> fullcontact sip:[email protected]:5060 >> lastms 0 >> useragent >> context default >> directmedia no >> deny 0.0.0.0/0.0.0.0 >> permit PU.BL.IC.IP >> nat force_rport,comedia >> language >> disallow >> allow >> force_avp yes >> callerid >> amaflags >> mailbox >> regexten >> regserver >> fromdomain testers.com >> videosupport no >> contactpermit >> contactdeny >> fullname 660 win8 >> hasvoicemail >> subscribemwi >> dtlsenable yes >> dtlsverify no >> dtlscertfile /etc/asterisk/keys/asterisk.pem >> dtlsprivatekey /etc/asterisk/keys/asterisk.pem >> dtlssetup actpass >> sippasswd md5pwd >> rpid >> domain testers.com >> sippasswd2 >> >> and my sip.conf: >> >> [general] >> bindport = 5070 >> bindaddr = PU.BL.IC.IP >> udpbindaddr = PU.BL.IC.IP >> tcpenable = yes >> limitonpeers = yes >> rtcachefriends = no >> tos_sip=cs3 >> tos_audio=ef >> realm = testers.com >> autodomain=yes >> domain=PU.BL.IC.IP >> domain=testers.com >> transport=ws,wss,udp >> outboundproxy=PU.BL.IC.IP:5060 >> >> >> I'd appreciate Your advice. >> >> >> > What does a DEBUG log show with 'sip set debug on' when the outbound call > is made? > > -- > Matthew Jordan > Digium, Inc. | Engineering Manager > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: http://digium.com & http://asterisk.org > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
