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Jonas Kellens

2016-Aug-10 20:01 UTC

[asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

Hello

thank you for your answer.

I don't understand how there are many tutorials and examples on the web 
where every time the outcome is a working setup. Very strange I feel now 
after my personal experience with Asterisk 11 and webRTC.

You also say Asterisk 13. How about Asterisk 12 then ??



Kind regards.



On 10-08-16 21:53, Matt Fredrickson wrote:> I don't see an ice-ufrag or ice-pwd line in the response from
> Asterisk, correlating with your suspicion that there is no ICE.  Are
> you sure that the stun server you're using (the google one) still
> works?  I haven't tried that server in a while, but I distantly seem
> to recall that maybe they shut it down.
>
> Asterisk 13 is a better place to be as well.  Asterisk 11 hasn't been
> feature updated in a while, and it could be that it could be a number
> of patches/fixes behind with regards to webrtc support, particularly
> with regards to interoperating with a modern browser version.
>
> Hope that helps,
> Matthew Fredrickson
>
> On Wed, Aug 10, 2016 at 5:02 AM, Jonas Kellens <jonas.kellens at
telenet.be> wrote:
>> On 10-08-16 08:52, Ludovic Gasc wrote:
>>
>> For WebRTC, I recommend you to use Asterisk 13+.
>>
>> Have a nice day.
>>
>> Ludovic Gasc (GMLudo)
>> http://www.gmludo.eu/
>>
>>
>>
>>
>> Hello
>>
>> then why is there an option in sip.conf and rtp.conf "
icesupport=yes" ??
>>
>> This is no answer to my question.
>>
>> So again : what am I missing to get ICE support on my Asterisk 11.23.0
??
>>
>>
>>
>> Kind regards.
>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- 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
>
>
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Matt Fredrickson

2016-Aug-10 20:03 UTC

head link

[asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

My suggestion is to verify and debug against Asterisk 13 first, and
then you can try backing down versions, rather than reverse.  WebRTC
is a rapidly moving target, and has required ongoing changes that may
not have made it into older and feature frozen versions of Asterisk.

Matthew Fredrickson

On Wed, Aug 10, 2016 at 3:01 PM, Jonas Kellens <jonas.kellens at
telenet.be> wrote:> Hello
>
> thank you for your answer.
>
> I don't understand how there are many tutorials and examples on the web
> where every time the outcome is a working setup. Very strange I feel now
> after my personal experience with Asterisk 11 and webRTC.
>
> You also say Asterisk 13. How about Asterisk 12 then ??
>
>
>
> Kind regards.
>
>
>
> On 10-08-16 21:53, Matt Fredrickson wrote:
>
> I don't see an ice-ufrag or ice-pwd line in the response from
> Asterisk, correlating with your suspicion that there is no ICE.  Are
> you sure that the stun server you're using (the google one) still
> works?  I haven't tried that server in a while, but I distantly seem
> to recall that maybe they shut it down.
>
> Asterisk 13 is a better place to be as well.  Asterisk 11 hasn't been
> feature updated in a while, and it could be that it could be a number
> of patches/fixes behind with regards to webrtc support, particularly
> with regards to interoperating with a modern browser version.
>
> Hope that helps,
> Matthew Fredrickson
>
> On Wed, Aug 10, 2016 at 5:02 AM, Jonas Kellens <jonas.kellens at
telenet.be>
> wrote:
>
> On 10-08-16 08:52, Ludovic Gasc wrote:
>
> For WebRTC, I recommend you to use Asterisk 13+.
>
> Have a nice day.
>
> Ludovic Gasc (GMLudo)
> http://www.gmludo.eu/
>
>
>
>
> Hello
>
> then why is there an option in sip.conf and rtp.conf "
icesupport=yes" ??
>
> This is no answer to my question.
>
> So again : what am I missing to get ICE support on my Asterisk 11.23.0 ??
>
>
>
> Kind regards.
>
>
>
> --
> _____________________________________________________________________
> -- 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


-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

Jonas Kellens

2016-Aug-11 14:09 UTC

head link

[asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

Hello

Using Asterisk 12.8.2.

I now have the "via ICE" messages in the RTP debug (see below).

If you look in the SIP debug (see below), you also now see the 
"ice-ufrag" and "ice-pwd" in the 200 OK SIP-message from
Asterisk to the
webRTC client.


But still no audio ! None at all ! In both directions.


You can see in the SIP debug that the IP-address in de SDP-body is 
correctly set for sending audio. So I don't think it is a NAT/ICE problem.


Can anyone tell me then what is left that could be causing the 
'no-audio' problem ??



SIP debug :


[Aug 11 15:53:47] <--- SIP read from WS:178.119.146.190:60191 --->
[Aug 11 15:53:47] INVITE sip:419 at 178.18.90.230 SIP/2.0
[Aug 11 15:53:47] Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;branch=z9hG4bKSqKu6K3uxr3dOFdU5WAtPM5tKKA5yzAq;rport
[Aug 11 15:53:47] From: 
<sip:770000wrtc at 178.18.90.230>;tag=SGUVL1LMdvxQrUfxprZJ
[Aug 11 15:53:47] To: <sip:419 at 178.18.90.230>
[Aug 11 15:53:47] Contact: 
<sips:770000wrtc at
df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr"
[Aug 11 15:53:47] Call-ID: 47ca4cc9-9dce-4449-d58f-e069a67061ec
[Aug 11 15:53:47] CSeq: 58874 INVITE
[Aug 11 15:53:47] Content-Type: application/sdp
[Aug 11 15:53:47] Content-Length: 2301
[Aug 11 15:53:47] Max-Forwards: 70
[Aug 11 15:53:47] Authorization: Digest 
username="770000wrtc",realm="178.18.90.230",nonce="5d5c700b",uri="sip:419
at
178.18.90.230",response="ca118222a4674b4c6dcc19dd95e00c15",algorithm=MD5
[Aug 11 15:53:47] User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
[Aug 11 15:53:47] Organization: Doubango Telecom
[Aug 11 15:53:47]
[Aug 11 15:53:47] v=0
[Aug 11 15:53:47] o=- 5876454736929512000 2 IN IP4 127.0.0.1
[Aug 11 15:53:47] s=Doubango Telecom - chrome
[Aug 11 15:53:47] t=0 0
[Aug 11 15:53:47] a=group:BUNDLE audio
[Aug 11 15:53:47] a=msid-semantic: WMS kBwsfCPizGNiVjZS23dGoTNcUubDMMhxXrka
[Aug 11 15:53:47] m=audio 63897 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 
105 13 126
[Aug 11 15:53:47] c=IN IP4 178.119.146.190
[Aug 11 15:53:47] a=rtcp:63899 IN IP4 178.119.146.190
[Aug 11 15:53:47] a=candidate:2999745851 1 udp 2122260223 192.168.56.1 
63896 typ host generation 0 network-id 1
[Aug 11 15:53:47] a=candidate:3378846520 1 udp 2122194687 192.168.1.120 
63897 typ host generation 0 network-id 2
[Aug 11 15:53:47] a=candidate:2999745851 2 udp 2122260222 192.168.56.1 
63898 typ host generation 0 network-id 1
[Aug 11 15:53:47] a=candidate:3378846520 2 udp 2122194686 192.168.1.120 
63899 typ host generation 0 network-id 2
[Aug 11 15:53:47] a=candidate:1210916236 1 udp 1685987071 
178.119.146.190 63897 typ srflx raddr 192.168.1.120 rport 63897 
generation 0 network-id 2
[Aug 11 15:53:47] a=candidate:1210916236 2 udp 1685987070 
178.119.146.190 63899 typ srflx raddr 192.168.1.120 rport 63899 
generation 0 network-id 2
[Aug 11 15:53:47] a=candidate:4233069003 1 tcp 1518280447 192.168.56.1 9 
typ host tcptype active generation 0 network-id 1
[Aug 11 15:53:47] a=candidate:2280056776 1 tcp 1518214911 192.168.1.120 
9 typ host tcptype active generation 0 network-id 2
[Aug 11 15:53:47] a=candidate:4233069003 2 tcp 1518280446 192.168.56.1 9 
typ host tcptype active generation 0 network-id 1
[Aug 11 15:53:47] a=candidate:2280056776 2 tcp 1518214910 192.168.1.120 
9 typ host tcptype active generation 0 network-id 2
[Aug 11 15:53:47] a=ice-ufrag:TxJQpv1i5O04Q+Kw
[Aug 11 15:53:47] a=ice-pwd:LvfUjrDPbY/np215T3+6Sy03
[Aug 11 15:53:47] a=fingerprint:sha-256 
EF:A4:78:E4:C1:33:5A:F5:36:6B:C5:DF:C7:D9:10:44:FD:96:5D:88:79:AB:8C:A0:E2:71:66:DA:6D:2C:30:84
[Aug 11 15:53:47] a=setup:actpass
[Aug 11 15:53:47] a=mid:audio
[Aug 11 15:53:47] a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
[Aug 11 15:53:47] a=extmap:3 
http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
[Aug 11 15:53:47] a=sendrecv
[Aug 11 15:53:47] a=rtcp-mux
[Aug 11 15:53:47] a=rtpmap:111 opus/48000/2
[Aug 11 15:53:47] a=rtcp-fb:111 transport-cc
[Aug 11 15:53:47] a=fmtp:111 minptime=10;useinbandfec=1
[Aug 11 15:53:47] a=rtpmap:103 ISAC/16000
[Aug 11 15:53:47] a=rtpmap:104 ISAC/32000
[Aug 11 15:53:47] a=rtpmap:9 G722/8000
[Aug 11 15:53:47] a=rtpmap:0 PCMU/8000
[Aug 11 15:53:47] a=rtpmap:8 PCMA/8000
[Aug 11 15:53:47] a=rtpmap:106 CN/32000
[Aug 11 15:53:47] a=rtpmap:105 CN/16000
[Aug 11 15:53:47] a=rtpmap:13 CN/8000
[Aug 11 15:53:47] a=rtpmap:126 telephone-event/8000
[Aug 11 15:53:47] a=ssrc:54412034 cname:H2asKiJklFa9L3Xw
[Aug 11 15:53:47] a=ssrc:54412034 
msid:kBwsfCPizGNiVjZS23dGoTNcUubDMMhxXrka 
f25030f2-3e48-4180-aea4-4edec3e67410
[Aug 11 15:53:47] a=ssrc:54412034 
mslabel:kBwsfCPizGNiVjZS23dGoTNcUubDMMhxXrka
[Aug 11 15:53:47] a=ssrc:54412034 label:f25030f2-3e48-4180-aea4-4edec3e67410
[Aug 11 15:53:47] <------------->
[Aug 11 15:53:47] --- (13 headers 44 lines) ---
[Aug 11 15:53:47] Using INVITE request as basis request - 
47ca4cc9-9dce-4449-d58f-e069a67061ec
[Aug 11 15:53:47] Found peer '770000wrtc' for '770000wrtc' from 
178.119.146.190:60191
[Aug 11 15:53:47]   == Using SIP RTP TOS bits 184
[Aug 11 15:53:47]   == Using SIP RTP CoS mark 5
[Aug 11 15:53:47] Found RTP audio format 111
[Aug 11 15:53:47] Found RTP audio format 103
[Aug 11 15:53:47] Found RTP audio format 104
[Aug 11 15:53:47] Found RTP audio format 9
[Aug 11 15:53:47] Found RTP audio format 0
[Aug 11 15:53:47] Found RTP audio format 8
[Aug 11 15:53:47] Found RTP audio format 106
[Aug 11 15:53:47] Found RTP audio format 105
[Aug 11 15:53:47] Found RTP audio format 13
[Aug 11 15:53:47] Found RTP audio format 126
[Aug 11 15:53:47] Found audio description format opus for ID 111
[Aug 11 15:53:47] Found unknown media description format ISAC for ID 103
[Aug 11 15:53:47] Found unknown media description format ISAC for ID 104
[Aug 11 15:53:47] Found audio description format G722 for ID 9
[Aug 11 15:53:47] Found audio description format PCMU for ID 0
[Aug 11 15:53:47] Found audio description format PCMA for ID 8
[Aug 11 15:53:47] Found unknown media description format CN for ID 106
[Aug 11 15:53:47] Found unknown media description format CN for ID 105
[Aug 11 15:53:47] Found audio description format CN for ID 13
[Aug 11 15:53:47] Found audio description format telephone-event for ID 126
[Aug 11 15:53:47] Capabilities: us - (alaw), peer - 
audio=(ulaw|alaw|g722|opus)/video=(nothing)/text=(nothing), combined - 
(alaw)
[Aug 11 15:53:47] Non-codec capabilities (dtmf): us - 0x1 
(telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 
(telephone-event|)
[Aug 11 15:53:47] Peer audio RTP is at port 178.119.146.190:63897
[Aug 11 15:53:47] Looking for 419 in testwebrtc (domain 178.18.90.230)
[Aug 11 15:53:47] list_route: route/path hop: 
<sips:770000wrtc at
df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>
[Aug 11 15:53:47]
[Aug 11 15:53:47] <--- Transmitting (NAT) to 178.119.146.190:60191 --->
[Aug 11 15:53:47] SIP/2.0 100 Trying
[Aug 11 15:53:47] Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;branch=z9hG4bKSqKu6K3uxr3dOFdU5WAtPM5tKKA5yzAq;received=178.119.146.190;rport=60191
[Aug 11 15:53:47] From: 
<sip:770000wrtc at 178.18.90.230>;tag=SGUVL1LMdvxQrUfxprZJ
[Aug 11 15:53:47] To: <sip:419 at 178.18.90.230>
[Aug 11 15:53:47] Call-ID: 47ca4cc9-9dce-4449-d58f-e069a67061ec
[Aug 11 15:53:47] CSeq: 58874 INVITE
[Aug 11 15:53:47] Server: myPBX
[Aug 11 15:53:47] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, 
SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Aug 11 15:53:47] Supported: replaces
[Aug 11 15:53:47] Contact: <sip:419 at 178.18.90.230:5060;transport=WS>
[Aug 11 15:53:47] Content-Length: 0
[Aug 11 15:53:47]
[Aug 11 15:53:47]
[Aug 11 15:53:47] <------------>
[Aug 11 15:53:47]   == Using SIP RTP TOS bits 184
[Aug 11 15:53:47]   == Using SIP RTP CoS mark 5
[Aug 11 15:53:47]     -- Called SIP/testacc7700905
[Aug 11 15:53:48]     -- SIP/testacc7700905-00000001 is ringing
[Aug 11 15:53:48]
[Aug 11 15:53:48] <--- Transmitting (NAT) to 178.119.146.190:60191 --->
[Aug 11 15:53:48] SIP/2.0 180 Ringing
[Aug 11 15:53:48] Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;branch=z9hG4bKSqKu6K3uxr3dOFdU5WAtPM5tKKA5yzAq;received=178.119.146.190;rport=60191
[Aug 11 15:53:48] From: 
<sip:770000wrtc at 178.18.90.230>;tag=SGUVL1LMdvxQrUfxprZJ
[Aug 11 15:53:48] To: <sip:419 at 178.18.90.230>;tag=as6a3f0437
[Aug 11 15:53:48] Call-ID: 47ca4cc9-9dce-4449-d58f-e069a67061ec
[Aug 11 15:53:48] CSeq: 58874 INVITE
[Aug 11 15:53:48] Server: myPBX
[Aug 11 15:53:48] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, 
SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Aug 11 15:53:48] Supported: replaces
[Aug 11 15:53:48] Contact: <sip:419 at 178.18.90.230:5060;transport=WS>
[Aug 11 15:53:48] Content-Length: 0
[Aug 11 15:53:48]
[Aug 11 15:53:48]
[Aug 11 15:53:48] <------------>
[Aug 11 15:53:50]        > 0x7f2d8c018ee0 -- Probation passed - setting 
RTP source address to 178.119.146.190:58814
[Aug 11 15:53:50] NOTICE[8910][C-00000000]: res_rtp_asterisk.c:4467 
ast_rtp_read: Unknown RTP codec 95 received from '178.119.146.190:58814'
[Aug 11 15:53:50]     -- SIP/testacc7700905-00000001 answered 
SIP/770000wrtc-00000000
[Aug 11 15:53:50] Audio is at 11780
[Aug 11 15:53:50] Adding codec 100004 (alaw) to SDP
[Aug 11 15:53:50] Adding non-codec 0x1 (telephone-event) to SDP
[Aug 11 15:53:50]
[Aug 11 15:53:50] <--- Reliably Transmitting (NAT) to 
178.119.146.190:60191 --->
[Aug 11 15:53:50] SIP/2.0 200 OK
[Aug 11 15:53:50] Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;branch=z9hG4bKSqKu6K3uxr3dOFdU5WAtPM5tKKA5yzAq;received=178.119.146.190;rport=60191
[Aug 11 15:53:50] From: 
<sip:770000wrtc at 178.18.90.230>;tag=SGUVL1LMdvxQrUfxprZJ
[Aug 11 15:53:50] To: <sip:419 at 178.18.90.230>;tag=as6a3f0437
[Aug 11 15:53:50] Call-ID: 47ca4cc9-9dce-4449-d58f-e069a67061ec
[Aug 11 15:53:50] CSeq: 58874 INVITE
[Aug 11 15:53:50] Server: myPBX
[Aug 11 15:53:50] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, 
SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Aug 11 15:53:50] Supported: replaces
[Aug 11 15:53:50] Contact: <sip:419 at 178.18.90.230:5060;transport=WS>
[Aug 11 15:53:50] Content-Type: application/sdp
[Aug 11 15:53:50] Content-Length: 969
[Aug 11 15:53:50]
[Aug 11 15:53:50] v=0
[Aug 11 15:53:50] o=myPBX 794545698 794545698 IN IP4 178.18.90.230
[Aug 11 15:53:50] s=myPBX
[Aug 11 15:53:50] c=IN IP4 178.18.90.230
[Aug 11 15:53:50] t=0 0
[Aug 11 15:53:50] m=audio 11780 UDP/TLS/RTP/SAVPF 8 126
[Aug 11 15:53:50] a=rtpmap:8 PCMA/8000
[Aug 11 15:53:50] a=rtpmap:126 telephone-event/8000
[Aug 11 15:53:50] a=fmtp:126 0-16
[Aug 11 15:53:50] a=ptime:20
[Aug 11 15:53:50] a=maxptime:150
[Aug 11 15:53:50] a=ice-ufrag:58a5f9de0d48369c30dba971059275db
[Aug 11 15:53:50] a=ice-pwd:0f085841667af68d2ebc1a055613d53e
[Aug 11 15:53:50] a=candidate:Hb21259ee 1 UDP 2130706431 178.18.90.230 
11780 typ host
[Aug 11 15:53:50] a=candidate:Ha0a0101 1 UDP 2130706431 10.10.1.1 11780 
typ host
[Aug 11 15:53:50] a=candidate:Sb21259ee 1 UDP 1694498815 178.18.90.230 
11780 typ srflx raddr 178.18.90.230 rport 11780
[Aug 11 15:53:50] a=candidate:Hb21259ee 2 UDP 2130706430 178.18.90.230 
11781 typ host
[Aug 11 15:53:50] a=candidate:Ha0a0101 2 UDP 2130706430 10.10.1.1 11781 
typ host
[Aug 11 15:53:50] a=candidate:Sb21259ee 2 UDP 1694498814 178.18.90.230 
11781 typ srflx raddr 178.18.90.230 rport 11781
[Aug 11 15:53:50] a=connection:new
[Aug 11 15:53:50] a=setup:active
[Aug 11 15:53:50] a=fingerprint:SHA-256 
DB:10:AC:29:28:3A:55:7A:68:59:57:3C:22:ED:C8:20:4F:79:CC:4E:01:F5:55:10:3D:B4:D2:DD:5B:24:1E:2A
[Aug 11 15:53:50] a=sendrecv
[Aug 11 15:53:50]
[Aug 11 15:53:50] <------------>
[Aug 11 15:53:50]     -- Channel SIP/770000wrtc-00000000 joined 
'simple_bridge' basic-bridge
<eca525df-711a-4f04-a575-71ce917b49e4>
[Aug 11 15:53:50]     -- Channel SIP/testacc7700905-00000001 joined 
'simple_bridge' basic-bridge
<eca525df-711a-4f04-a575-71ce917b49e4>
[Aug 11 15:53:50]
[Aug 11 15:53:50] <--- SIP read from WS:178.119.146.190:60191 --->
[Aug 11 15:53:50] ACK sip:419 at 178.18.90.230:5060;transport=WS SIP/2.0
[Aug 11 15:53:50] Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;branch=z9hG4bKosvPUfE7SGqs3pZo6muw;rport
[Aug 11 15:53:50] From: 
<sip:770000wrtc at 178.18.90.230>;tag=SGUVL1LMdvxQrUfxprZJ
[Aug 11 15:53:50] To: <sip:419 at 178.18.90.230>;tag=as6a3f0437
[Aug 11 15:53:50] Contact: 
<sips:770000wrtc at
df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr"
[Aug 11 15:53:50] Call-ID: 47ca4cc9-9dce-4449-d58f-e069a67061ec
[Aug 11 15:53:50] CSeq: 58874 ACK
[Aug 11 15:53:50] Content-Length: 0
[Aug 11 15:53:50] Max-Forwards: 70
[Aug 11 15:53:50] Authorization: Digest 
username="770000wrtc",realm="178.18.90.230",nonce="5d5c700b",uri="sip:419
at
178.18.90.230:5060;transport=WS",response="426b1c5b355ea70b9d23e3f5af161681",algorithm=MD5
[Aug 11 15:53:50] User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
[Aug 11 15:53:50] Organization: Doubango Telecom



RTP debug :


RTP Debugging Enabled for address: 178.119.146.190:0
[Aug 11 16:01:20] Got  RTP packet from    178.119.146.190:58814 (type 
08, seq 014114, ts 3292374327, len 000160)
[Aug 11 16:01:20] Sent RTP packet to      178.119.146.190:50760 (via 
ICE) (type 08, seq 033787, ts 3292374320, len 000160)
[Aug 11 16:01:20] Got  RTP packet from    178.119.146.190:58814 (type 
08, seq 014115, ts 3292374487, len 000160)
[Aug 11 16:01:20] Sent RTP packet to      178.119.146.190:50760 (via 
ICE) (type 08, seq 033788, ts 3292374480, len 000160)
[Aug 11 16:01:20] Got  RTP packet from    178.119.146.190:58814 (type 
08, seq 014116, ts 3292374647, len 000160)
[Aug 11 16:01:20] Sent RTP packet to      178.119.146.190:50760 (via 
ICE) (type 08, seq 033789, ts 3292374640, len 000160)
[Aug 11 16:01:20] Got  RTP packet from    178.119.146.190:58814 (type 
08, seq 014117, ts 3292374807, len 000160)
[Aug 11 16:01:20] Sent RTP packet to      178.119.146.190:50760 (via 
ICE) (type 08, seq 033790, ts 3292374800, len 000160)
[Aug 11 16:01:20] Got  RTP packet from    178.119.146.190:58814 (type 
08, seq 014118, ts 3292374967, len 000160)
[Aug 11 16:01:20] Sent RTP packet to      178.119.146.190:50760 (via 
ICE) (type 08, seq 033791, ts 3292374960, len 000160)
[Aug 11 16:01:20] Got  RTP packet from    178.119.146.190:58814 (type 
08, seq 014119, ts 3292375127, len 000160)
[Aug 11 16:01:20] Sent RTP packet to      178.119.146.190:50760 (via 
ICE) (type 08, seq 033792, ts 3292375120, len 000160)
[Aug 11 16:01:20] Got  RTP packet from    178.119.146.190:58814 (type 
08, seq 014120, ts 3292375287, len 000160)
[Aug 11 16:01:20] Sent RTP packet to      178.119.146.190:50760 (via 
ICE) (type 08, seq 033793, ts 3292375280, len 000160)
[Aug 11 16:01:20] Got  RTP packet from    178.119.146.190:58814 (type 
08, seq 014121, ts 3292375447, len 000160)
[Aug 11 16:01:20] Sent RTP packet to      178.119.146.190:50760 (via 
ICE) (type 08, seq 033794, ts 3292375440, len 000160)
[Aug 11 16:01:20] Got  RTP packet from    178.119.146.190:58814 (type 
08, seq 014122, ts 3292375607, len 000160)
[Aug 11 16:01:20] Sent RTP packet to      178.119.146.190:50760 (via 
ICE) (type 08, seq 033795, ts 3292375600, len 000160)
[Aug 11 16:01:20] Got  RTP packet from    178.119.146.190:58814 (type 
08, seq 014123, ts 3292375767, len 000160)
[Aug 11 16:01:20] Sent RTP packet to      178.119.146.190:50760 (via 
ICE) (type 08, seq 033796, ts 3292375760, len 000160)
[Aug 11 16:01:20] Got  RTP packet from    178.119.146.190:58814 (type 
08, seq 014124, ts 3292375927, len 000160)
[Aug 11 16:01:20] Sent RTP packet to      178.119.146.190:50760 (via 
ICE) (type 08, seq 033797, ts 3292375920, len 000160)
[Aug 11 16:01:20] Got  RTP packet from    178.119.146.190:58814 (type 
08, seq 014125, ts 3292376087, len 000160)
[Aug 11 16:01:20] Sent RTP packet to      178.119.146.190:50760 (via 
ICE) (type 08, seq 033798, ts 3292376080, len 000160)
[Aug 11 16:01:20] Got  RTP packet from    178.119.146.190:58814 (type 
08, seq 014126, ts 3292376247, len 000160)
[Aug 11 16:01:20] Sent RTP packet to      178.119.146.190:50760 (via 
ICE) (type 08, seq 033799, ts 3292376240, len 000160)
[Aug 11 16:01:20] Got  RTP packet from    178.119.146.190:58814 (type 
08, seq 014127, ts 3292376407, len 000160)
[Aug 11 16:01:20] Sent RTP packet to      178.119.146.190:50760 (via 
ICE) (type 08, seq 033800, ts 3292376400, len 000160)
[Aug 11 16:01:20] Got  RTP packet from    178.119.146.190:58814 (type 
08, seq 014128, ts 3292376567, len 000160)






On 10-08-16 22:03, Matt Fredrickson wrote:> My suggestion is to verify and debug against Asterisk 13 first, and
> then you can try backing down versions, rather than reverse.  WebRTC
> is a rapidly moving target, and has required ongoing changes that may
> not have made it into older and feature frozen versions of Asterisk.
>
> Matthew Fredrickson
>
> On Wed, Aug 10, 2016 at 3:01 PM, Jonas Kellens <jonas.kellens at
telenet.be> wrote:
>> Hello
>>
>> thank you for your answer.
>>
>> I don't understand how there are many tutorials and examples on the
web
>> where every time the outcome is a working setup. Very strange I feel
now
>> after my personal experience with Asterisk 11 and webRTC.
>>
>> You also say Asterisk 13. How about Asterisk 12 then ??
>>
>>
>>
>> Kind regards.
>>
>>
>>
>> On 10-08-16 21:53, Matt Fredrickson wrote:
>>
>> I don't see an ice-ufrag or ice-pwd line in the response from
>> Asterisk, correlating with your suspicion that there is no ICE.  Are
>> you sure that the stun server you're using (the google one) still
>> works?  I haven't tried that server in a while, but I distantly
seem
>> to recall that maybe they shut it down.
>>
>> Asterisk 13 is a better place to be as well.  Asterisk 11 hasn't
been
>> feature updated in a while, and it could be that it could be a number
>> of patches/fixes behind with regards to webrtc support, particularly
>> with regards to interoperating with a modern browser version.
>>
>> Hope that helps,
>> Matthew Fredrickson
>>
>> On Wed, Aug 10, 2016 at 5:02 AM, Jonas Kellens <jonas.kellens at
telenet.be>
>> wrote:
>>
>> On 10-08-16 08:52, Ludovic Gasc wrote:
>>
>> For WebRTC, I recommend you to use Asterisk 13+.
>>
>> Have a nice day.
>>
>> Ludovic Gasc (GMLudo)
>> http://www.gmludo.eu/
>>
>>
>>
>>
>> Hello
>>
>> then why is there an option in sip.conf and rtp.conf "
icesupport=yes" ??
>>
>> This is no answer to my question.
>>
>> So again : what am I missing to get ICE support on my Asterisk 11.23.0
??
>>
>>
>>
>> Kind regards.
>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- 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
>
>

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