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[ohjelmistoarkkitehti a...]

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:660@PU.BL.IC.IP: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.


cheers,
Olli

[mjordan at digium.com]

On Mon, Sep 8, 2014 at 9:48 AM, Olli Heiskanen <ohjelmistoarkkitehti@gmail.com (ohjelmistoarkkitehti@gmail.com)> wrote:

[ohjelmistoarkkitehti a...]

Hi Matthew,

Here's the debug output: 










<--- SIP read from UDP:PU.BL.IC.IP:5060 --->
INVITE sip:661@testers.com ([email]sip%3A661@testers.com[/email]) 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:661@testers.com ([email]sip%3A661@testers.com[/email])>
From: "660" <sip:660@testers.com ([email]sip%3A660@testers.com[/email])>;tag=856i7ei98p
Call-ID: oc0ppijresm05k2emsgt
CSeq: 3394 INVITE
Contact: <sip:660@testers.com ([email]sip%3A660@testers.com[/email]);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:660@testers.com ([email]sip%3A660@testers.com[/email])>;tag=856i7ei98p
To: <sip:661@testers.com ([email]sip%3A661@testers.com[/email])>
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:661@PU.BL.IC.IP: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:661@PU.BL.IC.IP:5060 SIP/2.0
Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK4e738afa;rport
Max-Forwards: 70
From: "660 win8" <sip:660@testers.com ([email]sip%3A660@testers.com[/email])>;tag=as73376885
To: <sip:661@PU.BL.IC.IP:5060>
Contact: <sip:660@PU.BL.IC.IP:5070>
Call-ID: 2f70cc9567be50a46ba2879d4391a7dc@testers.com (2f70cc9567be50a46ba2879d4391a7dc@testers.com)
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:660@testers.com ([email]sip%3A660@testers.com[/email])>;tag=as73376885
To: <sip:661@PU.BL.IC.IP:5060>
Call-ID: 2f70cc9567be50a46ba2879d4391a7dc@testers.com (2f70cc9567be50a46ba2879d4391a7dc@testers.com)
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:660@testers.com ([email]sip%3A660@testers.com[/email])>;tag=856i7ei98p
To: <sip:661@testers.com ([email]sip%3A661@testers.com[/email])>;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:661@PU.BL.IC.IP: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:660@testers.com ([email]sip%3A660@testers.com[/email])>;tag=as73376885
To: <sip:661@PU.BL.IC.IP:5060>;tag=b552f34cdfad88fd2d6dc20c55c3a3ed-ba78
Call-ID: 2f70cc9567be50a46ba2879d4391a7dc@testers.com (2f70cc9567be50a46ba2879d4391a7dc@testers.com)
CSeq: 102 INVITE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
Transmitting (NAT) to PU.BL.IC.IP:5060:
ACK sip:661@PU.BL.IC.IP:5060 SIP/2.0
Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK4e738afa;rport
Max-Forwards: 70
From: "660 win8" <sip:660@testers.com ([email]sip%3A660@testers.com[/email])>;tag=as73376885
To: <sip:661@PU.BL.IC.IP:5060>;tag=b552f34cdfad88fd2d6dc20c55c3a3ed-ba78
Contact: <sip:660@PU.BL.IC.IP:5070>
Call-ID: 2f70cc9567be50a46ba2879d4391a7dc@testers.com (2f70cc9567be50a46ba2879d4391a7dc@testers.com)
CSeq: 102 ACK
User-Agent: I Am the Devil
Content-Length: 0




---
Scheduling destruction of SIP dialog '2f70cc9567be50a46ba2879d4391a7dc@testers.com (2f70cc9567be50a46ba2879d4391a7dc@testers.com)' 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:660@testers.com ([email]sip%3A660@testers.com[/email])>;tag=856i7ei98p
To: <sip:661@testers.com ([email]sip%3A661@testers.com[/email])>;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:661@testers.com ([email]sip%3A661@testers.com[/email]) SIP/2.0
Via: SIP/2.0/UDP PU.BL.IC.IP;branch=z9hG4bK3cb9.e83bde7c484eec5374f679e7232435af.0
Max-Forwards: 69
To: <sip:661@testers.com ([email]sip%3A661@testers.com[/email])>;tag=as4298ec2e
From: "660" <sip:660@testers.com ([email]sip%3A660@testers.com[/email])>;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 <mjordan@digium.com (mjordan@digium.com)>:

[mjordan at digium.com]

On Mon, Sep 8, 2014 at 10:19 AM, Olli Heiskanen <ohjelmistoarkkitehti@gmail.com (ohjelmistoarkkitehti@gmail.com)> wrote:

[ohjelmistoarkkitehti a...]

Hi,


My bad, below is the output for peer 661 and the log output, which does not tell me much. I guess Asterisk assumes it's working correctly as there are no errors etc, however for some reason the INVITE leaves Asterisk without the ice definitions in the sdp. I must assume it's a configuration issue as I've been modifying the sip peer table recently, and installed Asterisk 12, then moved back to 11. I compiled the current 11.11.0 version from source, starting from ./configure --with-crypto --with-ssl --with-srtp, make menuselect etc.


*CLI> realtime load sippeers name 661
                   Column Name  Column Value
          --------------------  --------------------
                            id  6
                          type  friend
                          name  661
                          host  dynamic
                        secret
                    encryption  yes
                          avpf  yes
                    icesupport  yes
                        ipaddr  PU.BL.IC.IP
                          port  5060
                    regseconds  1410190721
                   defaultuser  661
                   fullcontact  sip:661@PU.BL.IC.IP: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  661 win8 minipc
                  hasvoicemail
                  subscribemwi
                    dtlsenable  yes
                    dtlsverify  no
                  dtlscertfile  /etc/asterisk/keys/asterisk.pem
                dtlsprivatekey  /etc/asterisk/keys/asterisk.pem
                     dtlssetup  actpass
                     sippasswd  f825851fe6805899cb141bd469457829
                          rpid
                        domain  testers.com
                    sippasswd2







[Sep  8 21:18:12] VERBOSE[9315][C-00000002] netsock2.c:   == Using SIP RTP TOS bits 184
[Sep  8 21:18:12] VERBOSE[9315][C-00000002] netsock2.c:   == Using SIP RTP CoS mark 5
[Sep  8 21:18:12] VERBOSE[9406][C-00000002] pbx.c:     -- Executing [661@default:1] NoOp("SIP/660-00000002", "general : Dialed 661") in new stack
[Sep  8 21:18:12] VERBOSE[9406][C-00000002] pbx.c:     -- Executing [661@default:2] Dial("SIP/660-00000002", "SIP/661,3600,rt") in new stack
[Sep  8 21:18:12] VERBOSE[9406][C-00000002] netsock2.c:   == Using SIP RTP TOS bits 184
[Sep  8 21:18:12] VERBOSE[9406][C-00000002] netsock2.c:   == Using SIP RTP CoS mark 5
[Sep  8 21:18:12] VERBOSE[9406][C-00000002] app_dial.c:     -- Called SIP/661
[Sep  8 21:18:13] VERBOSE[9406][C-00000002] app_dial.c:   == Everyone is busy/congested at this time (1:0/0/1)
[Sep  8 21:18:13] VERBOSE[9406][C-00000002] pbx.c:     -- Executing [661@default:3] Hangup("SIP/660-00000002", "") in new stack
[Sep  8 21:18:13] VERBOSE[9406][C-00000002] pbx.c:   == Spawn extension (default, 661, 3) exited non-zero on 'SIP/660-00000002'








Here is the log output of my 661 sip.js client:
There the client receives an INVITE without ice data in the sdp, complains about it and responds with a 488. 




Mon Sep 08 2014 20:21:41 GMT+0300 (FLE Daylight Time) | sip.transport | received WebSocket text message:


INVITE sip:lgvt0hci@lqne1q8dttn3.invalid;transport=ws SIP/2.0
Record-Route: <sip:PU.BL.IC.IP;transport=ws;r2=on;lr=on;ftag=as40c12073>
Record-Route: <sip:PU.BL.IC.IP;r2=on;lr=on;ftag=as40c12073>
Via: SIP/2.0/WS  PU.BL.IC.IP;branch=z9hG4bKff35.a9b596f609aadeccdf58183a9fe4fcba.1
Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK39a2221b;rport=5070
Max-Forwards: 69
From: "660 win8" <sip:660@testers.com ([email]sip%3A660@testers.com[/email])>;tag=as40c12073
To: <sip:661@PU.BL.IC.IP:5060>
Contact: <sip:660@PU.BL.IC.IP:5070>
Call-ID: 597260a76cb0cb9155392f3a3c0be217@testers.com (597260a76cb0cb9155392f3a3c0be217@testers.com)
CSeq: 102 INVITE
User-Agent: I Am the Devil
Date: Mon, 08 Sep 2014 17:21:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 439


v=0
o=root 1283889088 1283889088 IN IP4 PU.BL.IC.IP
s=Asterisk PBX 11.11.0
c=IN IP4 PU.BL.IC.IP
t=0 0
m=audio 16822 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


...


Mon Sep 08 2014 20:21:41 GMT+0300 (FLE Daylight Time) | sip.transport | sending WebSocket message:


SIP/2.0 100 Trying
Via: SIP/2.0/WS  PU.BL.IC.IP;branch=z9hG4bKff35.a9b596f609aadeccdf58183a9fe4fcba.1
Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK39a2221b;rport=5070
To: <sip:661@PU.BL.IC.IP:5060>
From: "660 win8" <sip:660@testers.com ([email]sip%3A660@testers.com[/email])>;tag=as40c12073
Call-ID: 597260a76cb0cb9155392f3a3c0be217@testers.com (597260a76cb0cb9155392f3a3c0be217@testers.com)
CSeq: 102 INVITE
Supported: gruu,outbound
Content-Length: 0


...


Mon Sep 08 2014 20:21:41 GMT+0300 (FLE Daylight Time) | sip.inviteservercontext | invalid SDP sip-0.6.2.js:2655
Mon Sep 08 2014 20:21:41 GMT+0300 (FLE Daylight Time) | sip.inviteservercontext | Failed to set remote offer sdp: Called with SDP without ice-ufrag and ice-pwd. sip-0.6.2.js:2655
Mon Sep 08 2014 20:21:41 GMT+0300 (FLE Daylight Time) | sip.transport | sending WebSocket message:


SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/WS  PU.BL.IC.IP;branch=z9hG4bKff35.a9b596f609aadeccdf58183a9fe4fcba.0
Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK39a2221b;rport=5070
To: <sip:661@PU.BL.IC.IP:5060>;tag=cl8lmb52gl
From: "660 win8" <sip:660@testers.com ([email]sip%3A660@testers.com[/email])>;tag=as40c12073
Call-ID: 597260a76cb0cb9155392f3a3c0be217@testers.com (597260a76cb0cb9155392f3a3c0be217@testers.com)
CSeq: 102 INVITE
Supported: gruu,outbound
Content-Length: 0


...


Mon Sep 08 2014 20:21:41 GMT+0300 (FLE Daylight Time) | sip.transport | received WebSocket text message:


ACK sip:v1vbuq35@0i03dp4lli27.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS  PU.BL.IC.IP;branch=z9hG4bKff35.a9b596f609aadeccdf58183a9fe4fcba.0
Max-Forwards: 69
From: "660 win8" <sip:660@testers.com ([email]sip%3A660@testers.com[/email])>;tag=as40c12073
To: <sip:661@PU.BL.IC.IP:5060>;tag=cl8lmb52gl
Call-ID: 597260a76cb0cb9155392f3a3c0be217@testers.com (597260a76cb0cb9155392f3a3c0be217@testers.com)
CSeq: 102 ACK
Content-Length: 0










Thanks,
Olli






2014-09-08 18:50 GMT+03:00 Matthew Jordan <mjordan@digium.com (mjordan@digium.com)>: