[asterisk-users] Asterisk removes ice lines in sdp when calling between webrtc clients

Mon Sep 8 10:50:16 CDT 2014

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

> Hi Matthew,
>
> Here's the debug output:
>
>
>
>
>
> <--- SIP read from UDP:PU.BL.IC.IP:5060 --->
> INVITE sip:661 at testers.com 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 at testers.com>
> From: "660" <sip:660 at testers.com>;tag=856i7ei98p
> Call-ID: oc0ppijresm05k2emsgt
> CSeq: 3394 INVITE
> Contact: <sip:660 at testers.com
> ;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 at testers.com>;tag=856i7ei98p
> To: <sip:661 at testers.com>
> 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 at PU.BL.IC.IP:5070>
> Content-Length: 0
>
>
> <------------>
>     -- Executing [661 at default:1] NoOp("SIP/660-00000007", "general :
> Dialed 661") in new stack
>     -- Executing [661 at 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 at 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 at testers.com>;tag=as73376885
> To: <sip:661 at PU.BL.IC.IP:5060>
> Contact: <sip:660 at PU.BL.IC.IP:5070>
> Call-ID: 2f70cc9567be50a46ba2879d4391a7dc at 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
>
> ---
>
>
That's not really DEBUG output - just VERBOSE output from the CLI with 'sip
set debug on'.

That aside, your initial e-mail provided the configuration for SIP peer
660, but not for SIP peer 661. In your dialplan, you are dialling SIP peer
661:

    -- Executing [661 at 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

What is their configuration?

-- 
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
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