[asterisk-users] Asterisk removes ice lines in sdp when calling between webrtc clients
Olli Heiskanen
ohjelmistoarkkitehti at gmail.com
Mon Sep 8 10:19:59 CDT 2014
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
---
<--- 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 at testers.com>;tag=as73376885
To: <sip:661 at PU.BL.IC.IP:5060>
Call-ID: 2f70cc9567be50a46ba2879d4391a7dc at 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 at testers.com>;tag=856i7ei98p
To: <sip:661 at testers.com>;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 at 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 at testers.com>;tag=as73376885
To: <sip:661 at PU.BL.IC.IP:5060>;tag=b552f34cdfad88fd2d6dc20c55c3a3ed-ba78
Call-ID: 2f70cc9567be50a46ba2879d4391a7dc at testers.com
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Transmitting (NAT) to PU.BL.IC.IP:5060:
ACK 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>;tag=b552f34cdfad88fd2d6dc20c55c3a3ed-ba78
Contact: <sip:660 at PU.BL.IC.IP:5070>
Call-ID: 2f70cc9567be50a46ba2879d4391a7dc at testers.com
CSeq: 102 ACK
User-Agent: I Am the Devil
Content-Length: 0
---
Scheduling destruction of SIP dialog '
2f70cc9567be50a46ba2879d4391a7dc at testers.com' in 32000 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [661 at 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 at testers.com>;tag=856i7ei98p
To: <sip:661 at testers.com>;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 at testers.com SIP/2.0
Via: SIP/2.0/UDP
PU.BL.IC.IP;branch=z9hG4bK3cb9.e83bde7c484eec5374f679e7232435af.0
Max-Forwards: 69
To: <sip:661 at testers.com>;tag=as4298ec2e
From: "660" <sip:660 at testers.com>;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 at digium.com>:
>
>
> On Mon, Sep 8, 2014 at 9:48 AM, Olli Heiskanen <
> ohjelmistoarkkitehti at gmail.com> 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:660 at 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.
>>
>>
>>
> 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
>
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