thr3ads.net - asterisk users - [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ? [Aug 2016]

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Jonas Kellens

2016-Aug-11 14:09 UTC

[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
>>
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>> 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
>
>

Jonathan H

2016-Aug-11 14:25 UTC

head link

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

I'm genuinely fascinated why you are insisting on using a version of
Asterisk almost 3 years old, for which EOL support ended last year.

Is there any particular reason you cannot or will not use the current
version as others have suggested?

Also, I see you are using Doubango and WebRTC, but in the logs, I see WS
and WSS.

You NEED to be using 100% WSS otherwise you've not got a hope in hell of
anything working with WEBRTC.
Check the console of the web browser you are trying to make the call from
(CTRL-SHIFT-I in Chrome on Windows, for example).

Also, you'll need to be using valid certificates - self-signed certificates
won't work for any current implementation of WebRTC that I know of,
certainly not if anything involves current versions of Chrome or Firefox.
That said, LetsEncrypt certs work fine for this, so no need to spend out on
one.

Switch to Asterisk 13.10 and save yourself a whole lotta headache.

On 11 August 2016 at 15:09, Jonas Kellens <jonas.kellens at telenet.be>
wrote:
> Hello
>
> Using Asterisk 12.8.2.
>
>
> 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.
>
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Jonas Kellens

2016-Aug-11 14:40 UTC

head link

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

My main reason not to upgrade to Ast 13 is because I'm afraid of losing 
functionality as there are certain functions deprecated/replaced. This 
can also cause headache :-)

I will do so if there is no other option.

But still, I don't see why Ast 13 would differ so much in this case ? If 
ICE and NAT is working (not causing problems) why should Ast 13 bring me 
audio and Ast 12 don't ??



I indeed use SIPML5 demo as quick test-case. So do many tutorials on the 
web.

Self-signed certificates should be OK as long as they are imported in 
the browser. Never knew this could cause audio problems ?




Kind regards.



On 11-08-16 16:25, Jonathan H wrote:> I'm genuinely fascinated why you are insisting on using a version of 
> Asterisk almost 3 years old, for which EOL support ended last year.
>
> Is there any particular reason you cannot or will not use the current 
> version as others have suggested?
>
> Also, I see you are using Doubango and WebRTC, but in the logs, I see 
> WS and WSS.
>
> You NEED to be using 100% WSS otherwise you've not got a hope in hell 
> of anything working with WEBRTC.
> Check the console of the web browser you are trying to make the call 
> from (CTRL-SHIFT-I in Chrome on Windows, for example).
>
> Also, you'll need to be using valid certificates - self-signed 
> certificates won't work for any current implementation of WebRTC that 
> I know of, certainly not if anything involves current versions of 
> Chrome or Firefox. That said, LetsEncrypt certs work fine for this, so 
> no need to spend out on one.
>
> Switch to Asterisk 13.10 and save yourself a whole lotta headache.
>
> On 11 August 2016 at 15:09, Jonas Kellens <jonas.kellens at telenet.be 
> <mailto:jonas.kellens at telenet.be>> wrote:
>
>     Hello
>
>     Using Asterisk 12.8.2.
>
>
>     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.
>
>
>
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